Android Marshmallow AOSP Changes

Changes from 6.0.1_r62 (MTC20F) to 6.0.1_r63 (MXC89L):

Warning Releases with no significant changes other than version bump in platform/build component are likely to only feature proprietary binary blob (e.g. firmwares) changes.

Removed Components (0):

None

Updated Components (98):

  • device/asus/flo with 1 change(s)
    • e2fa6dc : Fix Image size for Razor and RazorG by reduce Jounal size to zero

  • device/asus/fugu-kernel with 2 change(s)
    • 724b4e2 : fugu: update prebuilt kernel
    • 4c12b58 : fugu: update prebuilt kernel

  • device/htc/flounder with 2 change(s)
    • 1f3dae7 : remove persist.sys.usb.config override
    • 251682c : Cherry pick for CL 698892.

  • device/htc/flounder-kernel with 1 change(s)

  • device/huawei/angler with 46 change(s)
    • 36e6067 : Modify touch boost to impact sched migration.
    • 1df99c4 : angler: thermal: allow thermal-engine daemon shutdown device when SELinux is enforcing
    • 0b083ef : angler: thermal: modify shutdown temperature threshold
    • 9d8866b : Allow ims create permission for netlink_route_socket
    • 2bedc24 : power-hal: Fix make file to use correct power hal library
    • c6e1f82 : power: Update hint action for camera hint
    • 3aaae04 : Add rules for device-services.
    • 3172d68 : Allow ims net_raw and net_admin permissions.
    • 76753dc : Enable low power video mode for 4K encode
    • a0b3046 : Revert "Revert "Enable scheduler boost for zygote.""
    • f1f0d98 : ARM: msm: angler: update the wifi nvram and add another two for calibration
    • f85a764 : adding nlmsg_read permission for WFC
    • e8c830e : angler: apns-full-conf: add Newco sim card to apn =pcweb.metropcs.com
    • ff5cfd8 : Angler: power: Update power consumption values.
    • 6dd07a0 : init.angler.power.sh: change GPU initial power level
    • ca35565 : angler: disable data toggle for powersaving.
    • 37a8681 : angler: Wifi: Config wifi hotspot channel(SW) for target customer
    • ec192f3 : angler: apns-full-conf: modify default APN to IPv4v6 according to AT&T Non-virtual apn
    • 7f8db63 : angler: init.angler.diag.rc.user: not remove /dev/diag in factory mode
    • 4d0fbd6 : angler: selinux: add selinux rules for atfwd
    • 60393b1 : angler: Sar control: Sar control service SELinux issue.
    • 56e3ebf : angler: apns-full-conf: delete the two APNs of Softbank requested
    • b07fc85 : angler: audio: improve audio parameters
    • 4410fbf : angler: Mul-PDP: Close Mul-PDP for special operators.
    • 929a30c : Revert "Move peripheral manger and proxy into enforcing mode."
    • 82acc27 : Make bluetooth state label more flexible.
    • 2d6cc58 : angler: Settings: Display FCC ID, IC ID, CE number, model number on the phone.
    • 8d1724d : angler: Settings: Display FCC ID, IC ID, CE number, model number on the phone.
    • e437e6e : angler: APN: No "ESM information transfer flag" IE in attach request message.
    • d29a759 : Fork apns-conf list to angler
    • c7cf855 : Add policy for qxdmlogger.
    • 9c7b319 : Move rild into permissive.
    • f4a5894 : Label /persist files and expand tee access.
    • 3f40ca7 : Relax timer_rate to 19000.
    • 90c3aa4 : Allow bluetooth access to bluetooth files.
    • 0dd3a64 : Enable small task packing for angler.
    • ae33944 : Lower hispeed_freq on Angler.
    • 5ee2c83 : Move tee into permissive mode.
    • e54bf02 : angler: HWC virtual display configuration
    • 40bceb1 : angler: Config: add operators requirement config for Sprint, Verizon, TMO and ATT.
    • dc064da : Label /dev/qseecom as tee_device.
    • c1db6dc : updated spi-contexthub.h kernel header
    • d57bb5f : location: update to AU170.
    • f018b26 : device.mk: add librmnetctl to PRODUCT_PACKAGES list.
    • a1b674c : angler: audio: change audio parameter for voice call with TMO sim card
    • 717587f : Revert "Enable file encryption on Angler"

  • device/huawei/angler-kernel with 24 change(s)
    • 333c445 : angler: update prebuilt kernel
    • 9764fd9 : angler: update prebuilt kernel
    • a6bab8c : angler: update prebuilt kernel
    • 234a1a2 : angler: update prebuilt kernel
    • bd08808 : angler: update prebuilt kernel
    • 4079682 : angler: update prebuilt kernel
    • 7f5e080 : angler: update prebuilt kernel
    • cedad8f : angler: update prebuilt kernel
    • b7db5f0 : angler: update prebuilt kernel
    • 2565c8d : angler: update prebuilt kernel
    • cb91e6b : angler: update prebuilt kernel
    • d9a4499 : angler: update prebuilt kernel
    • 6aaa4d9 : angler: update prebuilt kernel
    • 50634d1 : angler: update prebuilt kernel
    • 5d46205 : angler: update prebuilt kernel
    • 77b47e9 : angler: update prebuilt kernel
    • 5b0963f : angler: update prebuilt kernel
    • 037311b : angler: update prebuilt kernel
    • 3bf0d0c : angler: update prebuilt kernel AU199
    • 90b1765 : angler: update prebuilt kernel
    • df947d2 : angler: update prebuilt kernel
    • c97c24a : angler: update prebuilt kernel
    • 79aa487 : angler: update prebuilt kernel
    • f1f8351 : angler: update prebuilt kernel

  • device/lge/bullhead with 34 change(s)
    • 9bbfd3d : Modify touch boost to impact sched migration.
    • 80cd16c : Allow perfd write access to sched_upmigrate and sched_downmigrate
    • 425ea43 : bullhead/apns: Add MVNO apns for JP
    • e6956b0 : bullhead: Set mdc_initial_max_retry to 10 for Telstra
    • 78fb0a6 : bullhead: Set mdc_initial_max_retry to 10 for SBM
    • 96e5ff8 : bullhead: Wi-Fi ini update
    • d7d728b : power-hal: Fix make file to use correct power hal library
    • e96aecf : power: Update hint action for camera hint
    • 009fdf3 : Enable low power video mode for 4K encode
    • 58d1aaf : Disable zram for bullhead.
    • a0866aa : bullhead/apns: Update apn info
    • e9f9278 : bullhead: Use mcc-specific operator name list for China and Taiwan
    • ecd4248 : Revert "Revert "Enable scheduler boost for zygote.""
    • bf03054 : Enable additional firmware logs
    • b13c152 : bullhead: Wi-Fi QCA6174 ini file update to AU294
    • e9d9c6c : Bullhead: NFC: Configuration for FW 10.1.18
    • 24ba748 : bullhead/apns: Update apns-full-conf.xml
    • e8c17c8 : bullhead: Wi-Fi QCA6174 ini file update to AU245
    • 0632418 : SEAndroid: location policies update
    • 8eb0a89 : Relax timer_rate to 19000.
    • b3bf670 : Lower hispeed_freq for bullhead.
    • 7704fd0 : Enable small task packing for bullhead.
    • eb53197 : Allow mediaserver perfd search access.
    • c97c4ad : Enable scheduler boost for zygote.
    • d1a44b2 : bullhead/sepolicy: allow system_server to search mpctl_data_file
    • 0954a3d : bullhead/audio: set voice call volume step to 7
    • 5bd723f : Force block encrypt bullhead
    • b7b0523 : Revert "Enable file encryption on Bullhead"
    • fd8401d : Revert "Turn off file system checks to gather crashed file system images"
    • e7efd50 : Turn off file system checks to gather crashed file system images
    • 8f90ca0 : Revert "Enable file encryption on Bullhead"
    • 7c74c40 : updated spi-contexthub.h kernel header
    • cf686cf : Turn on zram for bullhead
    • 2c7cb3c : bullhead: Disable VT feature

  • device/lge/bullhead-kernel with 22 change(s)
    • 29dda3b : bullhead: update prebuilt kernel
    • 4dcbd2b : bullhead: update prebuilt kernel
    • ddb0903 : bullhead: update prebuilt kernel
    • ec566e4 : bullhead: update prebuilt kernel
    • cb83beb : bullhead: update prebuilt kernel
    • 2ee2ca8 : bullhead: update prebuilt kernel
    • d619568 : bullhead: update prebuilt kernel
    • 263b206 : bullhead: update prebuilt kernel
    • a735aee : bullhead: update prebuilt kernel
    • fff9c25 : bullhead: update prebuilt kernel
    • 0d3478c : bullhead: update prebuilt kernel
    • 0cf4992 : bullhead: update prebuilt kernel
    • 9429aef : bullhead: update prebuilt kernel
    • 49b14dd : bullhead: update prebuilt kernel
    • a50b717 : bullhead: update prebuilt kernel
    • 73827b1 : bullhead: update prebuilt kernel
    • 40b1b4b : bullhead: update prebuilt kernel
    • adc46a9 : bullhead: update prebuilt kernel
    • 45ede18 : bullhead: update prebuilt kernel AU199
    • 72ab03e : bullhead: update prebuilt kernel
    • b8fbf9f : bullhead: update prebuilt kernel
    • 7c2296b : bullhead: update prebuilt kernel

  • device/lge/hammerhead-kernel with 3 change(s)
    • 1545eca : hammerhead: update kernel prebuilt
    • 505bded : hammerhead: update kernel prebuilt
    • 62fa58a : hammerhead: update prebuilt kernel

  • device/moto/shamu with 1 change(s)
    • aef7bce : remove persist.sys.usb.config override

  • device/moto/shamu-kernel with 4 change(s)
    • d473b14 : shamu: update prebuilt kernel
    • 2925038 : shamu: update prebuilt kernel
    • c6c6124 : shamu: update prebuilt kernel
    • 4ef1775 : shamu: update prebuilt kernel

  • device/sample with 2 change(s)
    • d52feb0 : Adding APN for ATT AGMS Global (310-380)
    • c0b4a21 : Change APNs as request of operators

  • platform/art with 10 change(s)
    • 4d93433 : Add missing null check to String::ToCharArray
    • 4833277 : ART: Change UninitializedThis tracking in the verifier
    • 603b4c2 : Fix some java_lang_Class related moving GC bugs
    • c1956de : Fix compaction bug in Class_getDeclaredMethodsUnchecked
    • d6ec651 : ART: Sometimes even empty methods take forever to verify
    • c2d3221 : Quick: Abolish kMirOpCheckPart2.
    • 85336e3 : Fix broken checks in IsValidPartOfMemberNameUtf8Slow.
    • 38f2085 : Update the remaining input index of phis after deleting an input.
    • 3a01631 : ART: Fix loop information after dead code elimination
    • 06cb4a9 : Fix constructor access check through reflection

  • platform/bionic with 3 change(s)
    • 8a19e09 : Update timezone data to 2016a
    • c3351ea : Work around incorrect dt_needed entries
    • 4cb434d : Fix regression in crash reporting

  • platform/bootable/recovery with 6 change(s)
    • da64ac2 : Fix integer overflows in recovery procedure.
    • b750e65 : imgdiff: skip spurious gzip headers in image files
    • 0f2f6a7 : imgdiff: skip spurious gzip headers in image files
    • a200639 : uncrypt: Support file level encryption.
    • babcffa : Revert "Change init sequence to support file level encryption"
    • 0460f69 : Revert "Zero blocks before BLKDISCARD"

  • platform/build with 415 change(s)

  • platform/cts with 50 change(s)
    • accb996 : Fix build
    • 6b8eae4 : Remove unnecessary layout request, avoid obtaining wrong bounds rect
    • bf1875a : Add an option to use log saver for generated XML report
    • 1b0bd2c : [CtsVerfier] ScreenLockBoundKeys - use different key ID
    • 9edd486 : Bug: 23003511 Fix CTS: AudioManagerTest#testSoundEffects failing
    • 38a07ad : CTS runner fixes
    • b6ab3b2 : MIDI CTS: remove tests that pass null callback
    • 861fc41 : Revert "Bug: 23003511 CTS: AudioManagerTest#testSoundEffects failing"
    • 6ecbb88 : Update CTS for unsupported legacy ConnectivityManager APIs
    • f049e59 : Fix contacts provider cts tests
    • 954eefe : media: combine log to reduce the size
    • b999e79 : Add MIDI feature to MNC section
    • d173212 : Permissions: Fix Auth CTS tests to reflect changes.
    • 53b27bd : Verify uninstall using "pm list".
    • bdf0b8e : testExtractAlpha was asserting a bug
    • a479b2c : CTS: TelephonyProviderTest failing
    • 40db3d8 : media: use small frame size for resource manager test.
    • edd4ec4 : Update last public framework attribute to pass CTS test.
    • 03db252 : Revert "add more logging to monkey seed test"
    • d32ddbf : Add Ble test cases to BuildCtsTemporarilyKnownFailureList.
    • fcd0c82 : Add logging statements to debug hangs.
    • 6716917 : media: deliver report to host
    • 98a6bcd : videoperf: specify timeout for VideoEncoderDecoderTest
    • 8a0ebb8 : use locked orientation instead of "nosensor"
    • 5aafe29 : MediaSyncTest: use float for playback rate tolerance to keep consistent with others.
    • e11e244 : media: deliver test report to host.
    • e548f98 : CTS: Adds conditions to bypass FileSystemPermissionTest.testDeviceTreeCpuCurrent
    • 308abc6 : videoperf: avoid potential ArrayIndexOutOfBounds
    • 7cc7c32 : Use example.com for strict mode cts test.
    • 4b61d4c : Remove not relevant browser tests - APIs are gone
    • 71dba80 : Remove android.security.cts.SeccompBpfTest.
    • 71e5ed7 : A better shadow value to cover more devices, handle TV theme override.
    • 7dd1e17 : Respect hard restriction to minimum password length in CTS
    • 7bea994 : Use simple break strategy for testSetTextLong
    • a99d048 : Split CTS-DEQP plan into two plans
    • 039b00b : Fix up Looper preparation.
    • 69c85ad : Fix build
    • a01622e : Dismiss dialogs during indeterminate progress dialog CTS tests
    • 4eaca29 : Use dup2() to make message queue tests less flaky.
    • 22e6c1c : Fix Drawable, PopupWindow CTS breakages
    • ac23f11 : fix build
    • 6da731e : Remove WebGL conformance tests from CTS
    • b35b30c : Temp disables for deqp tests failing due to missing driver fixes.
    • 3762271 : Ensure JobScheduler Connectivity CTS reenables WiFI
    • c145421 : Temporarily disable flaky EGL tests while waiting for driver fixes
    • 71f03b6 : Failing android.content.ContentResolver tests to CTS staging
    • 84d24c0 : Add workaround for ContetxtImpl.setTheme(), verify test setup
    • c9bc142 : Hide more failing CTS-DEQP tests.
    • 76fc052 : Hide failing GLES31 texture.border_clamp tests.
    • 93ee255 : Add STOPSHIP to CTS-DEQP plan generation.

  • platform/dalvik with 1 change(s)
    • 48c3cda : Fix potential buffer overrun.

  • platform/development with 1 change(s)
    • 61c57fd : Setup flags to make SDK images provisioned.

  • platform/external/aac with 2 change(s)
    • 48fbcdf : Fix aacDecoder_drcExtractAndMap()
    • feb4450 : Fix stack corruption happening in aacDecoder_drcExtractAndMap()

  • platform/external/boringssl with 2 change(s)
    • ab88957 : Fix encoding bug in i2c_ASN1_INTEGER
    • 7bde24a : Remove support for mis-encoded PKCS#8 DSA keys.

  • platform/external/bouncycastle with 2 change(s)
    • fc47108 : GCMParameters: fix insecure tag size
    • 4f5a324 : Register DSA OID for KeyFactory not just Signature

  • platform/external/chromium-webview with 19 change(s)
    • f21e832 : WebView AOSP Drop Request - 44.0.2403.114
    • 818ef2f : WebView AOSP Drop Request - 44.0.2403.90
    • ba40585 : WebView AOSP Drop Request - 44.0.2403.85
    • 915bc46 : WebView AOSP Drop Request - 44.0.2403.64
    • 297ac26 : WebView AOSP Drop Request - 44.0.2403.54
    • 8064614 : WebView AOSP Drop Request - 44.0.2403.39
    • 55d1035 : Revert "Revert "WebView AOSP Drop Request - 44.0.2403.33""
    • 4c9f37e : Revert "WebView AOSP Drop Request - 44.0.2403.33"
    • ca301b8 : WebView AOSP Drop Request - 44.0.2403.33
    • d17ce7e : WebView AOSP Drop Request - 44.0.2403.13
    • eb73773 : WebView AOSP Drop Request - 44.0.2403.13
    • 7ca5c4d : Update AOSP Webview Apks
    • 684aaa8 : WebView AOSP Drop Request - 44.0.2399.3
    • 7711878 : WebView AOSP Drop Request - 44.0.2399.3
    • 94fe079 : Update WebViewGoogle to 44.0.2399.0
    • 27a628d : Update WebViewGoogle to 44.0.2399.0
    • 38b1eb9 : Update WebView to 44.0.2394.0.
    • 0aac98c : Update WebView to 44.0.2394.0.
    • f69654b : Update Webview packages for AOSP targets to 44.0.2370.2

  • platform/external/compiler-rt with 1 change(s)
    • 23a87c1 : Build the ASAN RTL without RTTI.

  • platform/external/conscrypt with 11 change(s)
    • 120175c : Use SSL_session_reused to check when a session was reused
    • 7247401 : Fix updateAAD when offset is not 0
    • 90b0c44 : OpenSSLCipher: multiple calls to updateAAD were ignored
    • 3e5b885 : OpenSSLCipher: reset AAD when necessary
    • b0e7a58 : Fix compilation with OpenSSL
    • 35c2503 : Fix error conditions in certificate/PKCS#7 reading
    • e5d26bc : NativeCrypto: special case for empty cipher list
    • 5b59e99 : OpenSSLKey: unsupported algorithm is an InvalidKeyException
    • 308eaf6 : Throw InvalidKeyException when keystore key malformed
    • e3834b1 : NativeCrypto: allow default exceptions
    • 72ea3d6 : NativeCrypto: not finding a key is not fatal

  • platform/external/deqp with 3 change(s)
    • 243bc61 : Remove infeasible M tests.
    • 498bfb2 : Remove tests infeasible for M.
    • 02b40d2 : Fix primitive bbox line verification regression.

  • platform/external/dhcpcd with 1 change(s)
    • 94a7bd8 : Improve length checks in DHCP Options parsing of dhcpcd.

  • platform/external/drm_hwcomposer with 15 change(s)
    • bf20636 : drm_hwcomposer: Always call PrepareFrame() for compositions
    • 0049694 : drm_hwcomposer: Save the atomic_test result between tests
    • 2b0cdf8 : drm_hwcomposer: Save the atomic_test result between tests
    • 6cc5fbf : drm_hwcomposer: only check the composition after a geometry change
    • 4d844a4 : drm_hwcomposer: Skip layers which aren't on-screen
    • d09d9ea : drm_hwcomposer: set blending mode to kPreMult for GL output
    • 5c867b3 : drm_hwcomposer: Allow for multiple transforms at once
    • 551a211 : drm_hwcomposer: Check the composition before sending to frame worker
    • 93eed9e : drm_hwcomposer: Split DrmDisplayCompositor::SquashAll()
    • 91e9310 : drm_hwcomposer: Add test_only mode to CommitFrame
    • f59eaf0 : drm_hwcomposer: Create mode blob on modeset queue
    • 8de1cd0 : drm_hwcomposer: during SquashAll, skip layers with kSourceNone
    • 6d3b661 : drm_hwcomposer: composite down to a primary plane after a timeout
    • c3c27e0 : drm_hwcomposer: avoid creating release fences for invalid OutputFd
    • 435be95 : drm_hwcomposer: add AutoLock to automatically handle pthread_mutex_lock

  • platform/external/flac with 1 change(s)
    • 08ea4eb : Avoid free-before-initialize vulnerability in heap

  • platform/external/icu with 1 change(s)
    • 7408a76 : Update timezone data to 2016a

  • platform/external/jhead with 1 change(s)
    • 9cbb652 : Fix possible out of bounds access

  • platform/external/libavc with 12 change(s)
    • 116309d : Decoder: Initialize first_pb_nal_in_pic for error slices
    • 666e3b9 : Decoder: Fix slice number increment for error clips
    • 919b19c : Decoder: Do not conceal slices with invalid SPS/PPS
    • b7e01e4 : Decoder: Initialize slice parameters before concealing error MBs
    • 2eb9899 : Decoder: Memset few structures to zero to handle error clips
    • e5eb73e : Fix slice params for interlaced video
    • 8f62961 : Decoder: Fix for handling invalid intra mode
    • c1c58d7 : Decoder: Set u1_long_term_reference_flag to 0 for error concealment
    • 93f1180 : Ensure ih264d_start_of_pic() is not repeated in ih264d_mark_err_slice_skip()
    • 1a12ed8 : Decoder: Fix stack underflow in CAVLC 4x4 parse functions
    • 1bff181 : Encoder: Fixed an issue in handling FPS greater than 60
    • 4e09125 : Fixed few issues seen in CTS tests

  • platform/external/libdrm with 4 change(s)
    • f4aaf7b : drm: add ZBC query methods for nouveau
    • ff04743 : drm: add flag support for address space alloc
    • 2941095 : drm: add DRM_NOUVEAU_GEM_MAP/UNMAP
    • ee1d48a : drm: add zcull info and zcull bind for nouveau

  • platform/external/libmpeg2 with 3 change(s)
    • 2eb3a61 : Fixed out of bound read in flush_bits
    • e2e9cd6 : Fixed stack buffer overflow
    • 1af7ccf : Fix for handling streams which resulted in negative num_mbs_left

  • platform/external/libunwind with 1 change(s)
    • ce727f2 : Fix incorrect check of bool returning function.

  • platform/external/libvpx with 1 change(s)
    • 260987c : Fix ParseElementHeader to support 0 payload elements

  • platform/external/okhttp with 1 change(s)
    • d8d0b08 : Fix for HttpURLConnection not always throwing SocketTimeoutException

  • platform/external/sepolicy with 6 change(s)
    • 34b6e1b : expose control over unpriv perf access to shell
    • 39818ba : Further restrict socket ioctls available to apps
    • 211ed2d : Remove generic socket access from untrusted processes
    • c1cb855 : camera: Add AIDL interface for CameraServiceProxy.
    • 9dc28cd : Allow MediaProvider to traverse /mnt/media_rw.
    • 324cdd6 : Fix sepolicy-analyze libc++.so loading issue w/CTS.

  • platform/external/sonivox with 2 change(s)
    • 91559d8 : Sonivox: add SafetyNet log.
    • 9b999a2 : Sonivox: sanity check numSamples.

  • platform/external/tinyalsa with 12 change(s)
    • caab548 : tinymix: Remove the unused parameter from tinymix_set_byte_ctl()
    • f481dc6 : tinymix: Support more that 512 bytes in byte control
    • b39bcb4 : mixer: add support for TLV based byte controls
    • 74f9a58 : tinyalsa: unable to set BYTE mixer
    • c90f8e2 : tinymix: Add support for setting/getting a binary control
    • 7a53049 : tinymix: Remove the unused parameter from tinymix_set_byte_ctl()
    • 16ee253 : tinymix: Support more that 512 bytes in byte control
    • 8dd560e : mixer: add support for TLV based byte controls
    • b22ae76 : tinyalsa: unable to set BYTE mixer
    • effa011 : tinymix: Add support for setting/getting a binary control
    • 5c9cb35 : Add pcm_get_poll_fd
    • 3886a87 : Export pcm_mmap_avail

  • platform/external/tremolo with 1 change(s)
    • eeadefa : Check partword is in range for # of partitions

  • platform/external/webrtc with 10460 change(s)
    • 04cb763 : Add tests for verifying transport feedback for audio and video.
    • fcfc804 : Eliminate defines in talk/
    • 3542013 : Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
    • 2734d77 : Remove assert which was incorrectly added to TcpPort::OnSentPacket.
    • 55674ff : Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
    • 31c8d2e : Update with new default boringssl no-aes cipher suites. Re-enable tests.
    • e5e0e57 : Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
    • 688e308 : Re-land: "Use an explicit identifier in Config"
    • 7307952 : Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
    • 268493a : Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
    • 35aae2e : Remove libfuzzer trybot from default trybot set.
    • ff2a635 : Add ramp-up tests for transport sequence number with and w/o audio.
    • 709513d : Delete remnants of non-square pixel support from cricket::VideoFrame.
    • beed828 : Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
    • 2d110be : Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
    • 8432e1f : Re-enable tests that failed under Linux_Msan.
    • fca54f4 : Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
    • 09d944f : Roll chromium_revision 346fea9..099be58 (369082:369139)
    • 306efad : Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
    • 292e192 : Add build_protobuf variable.
    • a276e73 : Clean the code for external denoiser.
    • 2f7dea1 : [rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way
    • ea8c0f6 : Fix capture ntp time issue introduced with r11187.
    • 365543d : Roll chromium_revision 131167b..346fea9 (368784:369082)
    • 25249d9 : Use an explicit identifier in Config
    • e591f93 : Storing raw audio sink for default audio track.
    • 6955870 : Convert channel counts to size_t.
    • 92e677a : [rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function
    • 5584bf4 : Make :rtc_base_approved a public dep of :rtc_base.
    • e84e96e : NetEq: Fix a typo in a comment
    • 36220ae : Slap deprecation notices on Pass methods
    • d20e651 : Fix test bug introduced in r11101.
    • 3e1cfa7 : Delete unused method webrtc::VideoRendererInterface::SetSize.
    • 3235a27 : Updated chromium/.gclient and sync_chromium.py to not ignore third_party/ffmpeg. Was forgotten in this CL: https://codereview.webrtc.org/1575913003/
    • 2845a02 : Remove unused enum RTPDirections.
    • 3842c5c : Wire-up BWE feedback for audio receive streams.
    • 6183de6 : Remove tools/refactoring.
    • 127782b : Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal.
    • 16979e3 : Update .gitignore
    • 67e94fb : Add unit test for stand-alone denoiser and fixed some bugs.
    • b2328d1 : Remove additional channel constraints when Beamforming is enabled in AudioProcessing
    • e93ad1b : Roll chromium_revision 8c958e0..131167b (368561:368784)
    • 2a34688 : Make Beamforming dynamically settable for Android platform builds
    • 2bc63a1 : clang-format audio_device/mac.
    • a7446d2 : Change DTLS default from 1.0 to 1.2 for webrtc.
    • f6c318e : Update API for Objective-C RTCMediaSource.
    • e799bad : Move Objective-C video renderers to webrtc/api/objc.
    • 8102879 : Update API for Objective-C RTCMediaStreamTrack.
    • a2c353f : Update API for Objective-C RTCStats.
    • 7e8145f : [rtp_rtcp] rtcp::Tmmbr moved into own file
    • 27ed3cc : SCTP: Stopped accepting SSRCs higher than max. Seems to fix asan-related crash.
    • a9a1d2a : H.264: Default flags and pulling in openh264 and ffmpeg.
    • 7823495 : Move RTCI420Frame to webrtc/api/objc/RTCVideoFrame with minor style changes.
    • fd99dea : Roll chromium_revision 42ab10e..8c958e0 (368534:368561)
    • ef3d805 : [rtp_rtcp] rtcp::Tmmbn moved into own file explicetly unchanged.
    • d36efeb : Roll chromium_revision e738b54..42ab10e (368533:368534)
    • 4de0037 : Roll chromium_revision 7d97c94..e738b54 (368514:368533)
    • 3c05e6c : Disable EndToEndTest.TransportSeqNumOnAudioAndVideo for Dr Memory.
    • daa8749 : Revert of Roll chromium_revision 7d97c94..951c006 (368514:368525) (patchset #1 id:1 of https://codereview.webrtc.org/1577573002/ )
    • db21f63 : fix GN build break on native_client
    • 6109fc1 : Roll chromium_revision 7d97c94..951c006 (368514:368525)
    • 0697db6 : Roll chromium_revision 8a15a7f..7d97c94 (368391:368514)
    • 684e995 : Disable 2 video tests which fail on DrMemoryFull
    • f475d36 : Properly handle different transports having different SSL roles.
    • 25702cb : Misc. small cleanups.
    • 5de688e : Roll chromium_revision ede5d4f..8a15a7f (368310:368391)
    • 49c454e : Cleaning neteq_unittest resource files.
    • f1685c7 : Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac
    • e74eef1 : Add CreateSend/ReceiveTransport() methods to CallTest.
    • 37ebcf0 : Reland "Add APK targets to build libjingle tests for Android." patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/
    • b71b4f0 : Update attributes to match gclibc's ansidecl.h
    • 004851c : Roll chromium_revision 32569c6..ede5d4f (368258:368310)
    • e1ca167 : Add tracing to NetEqImpl::GetAudio
    • ec80f03 : Check the mic volume only periodically on Mac.
    • fbeb97e : Fix clang warning in peerconnection_jni.cc
    • 59bac1a : Fix for stats updated twice when switching content type (realtime - screenshare). Add unittest.
    • 95ab30c : Roll chromium_revision 6dd04c2..32569c6 (368115:368258)
    • a2b1e03 : Disable AudioDeviceAPITest.MicrophoneVolumeTests on Linux.
    • 893505d : Adding unit test to ensure TURN server priorities are unique.
    • e5ba13b : Adding a way for a Java RtpSender to set a track without taking ownership.
    • ced8ec9 : Roll chromium_revision bd5949f..6dd04c2 (368055:368115)
    • bedc17b : Fixing integer underflow in FileAudioDevice (webrtc issue 4554)
    • 6938793 : vp9 tests: Adjust some parameters and re-enable the tests.
    • 6f5ca08 : Update API for Objective-C RTCMediaConstraints.
    • 9fea80f : Add audio streams to CallTest and a first A/V call test.
    • ecd21b4 : Add ImplementationName to SimulcastEncoderAdapter.
    • 01f364e : Remove always-on options in OveruseFrameDetector.
    • 30166cb : iOS stability improvement for device switching, including BT devices
    • 7776e78 : Remove unused methods in VideoCodingModule.
    • 3886fc8 : Use pointer to generated FEC packet.
    • 46ea3ce : AudioDeviceTest.StartPlayoutOnTwoInstances now verifies two active playing streams
    • a49ad98 : Roll chromium_revision 4662d4f..bd5949f (368042:368055)
    • cea7c2f : Replace manual casting to rvalue reference with calls to std::move
    • a46a4c9 : Roll chromium_revision 2a70cb1..4662d4f (367468:368042)
    • 1fe48a5 : Add implementation in metrics.h that uses atomic pointer.
    • 4331fcd : Remove duplicate code in SocketDispatcher
    • 44cc795 : Roll chromium_revision 4df108a..2a70cb1 (367307:367468)
    • 4f3c772 : Update default trybot names for changed Mac bots
    • 67e83d6 : Update API for Objective-C RTCSessionDescription.
    • 29d5e57 : Update API for Objective-C RTCIceCandidate.
    • 1aa6e4e : Update default trybot names for changed Mac bots
    • 335ecf5 : Disable VideoCaptureTest.Capabilities and CreateDelete fails on Mac
    • b680274 : Fix a flaky turnport test failure
    • 6b9ab92 : Cease all future TURN requests when a TURN refresh request fails for a given TURN port. This fixes an assert error in Turnport::OnSendStunPacket
    • 37389b4 : Don't delete an ICE connection until it has been pruned or timed out on writing in the case where it hasn't received anything yet. Deleting an ICE connection before it is pruned or timed out when it hasn't received anything yet leads to ICE connections being deleted before they have a chance to send a ping and receive a response. BUG=
    • e2976c8 : Remove DISABLED_ON_ macros.
    • 13f61df : Move fake-handle frame creation into test target.
    • 5237021 : Disabling CritSectTest on Windows DrMemory bot.
    • 60ca31b : Roll chromium_revision d66326c..4df108a (367167:367307)
    • 4138e08 : Convert DOS-Unix line endings for Dr memory exclude file.
    • 06689a1 : Fix a -Wunused-function warning in gn builds.
    • 112fe43 : Fill the remote pwd in the ice candidates when an ICE credential is received. Also when a STUN ping arrives from an unknown address, try to find the pwd and generation from the remote ICE parameters.
    • 051b620 : Roll chromium_revision 58e631b..d66326c (367148:367167)
    • 41d1a62 : Use getExternalStorageDirectory() for trace file.
    • 555ad27 : Roll chromium_revision c670faf..58e631b (367136:367148)
    • 78ad5af : Roll chromium_revision 55b9604..c670faf (367087:367136)
    • 0c7e9f5 : Removing webrtc::PortAllocatorFactoryInterface.
    • e86e15b : Increasing timeout for TestResolverShutdown.
    • 355113a : Roll chromium_revision f84c240..55b9604 (367060:367087)
    • 831afda : Roll chromium_revision 75c7843..f84c240 (366966:367060)
    • 2df2ba7 : [rtp_rtcp] Fix CL#1539423003 public function RtpHeaderParser::Parse with old signature restored as deprecated.
    • 3f7219b : Fixing issue where description contains empty ICE ufrag/pwd.
    • 9faf154 : Reland 1531763006
    • 6d3f11c : Roll gtest-parallel.
    • f6975f4 : [rtp_rtcp] Lint errors cleaned from rtp_utility
    • e0d56a7 : Roll chromium_revision a97fe0c..75c7843 (366958:366966)
    • 63193ce : Roll chromium_revision a54fe37..a97fe0c (366930:366958)
    • 445e315 : Roll chromium_revision 083ad1e..a54fe37 (366925:366930)
    • 0e9f06f : Roll chromium_revision e73d852..083ad1e (366918:366925)
    • 74d8a11 : Roll chromium_revision e13537f..e73d852 (366904:366918)
    • bbff0d3 : Roll chromium_revision 0f24b97..e13537f (366898:366904)
    • 40a2f03 : Roll chromium_revision c80b1b6..0f24b97 (366886:366898)
    • ef776df : Roll chromium_revision 353749a..c80b1b6 (366864:366886)
    • 55e6159 : Roll chromium_revision 244de00..353749a (366862:366864)
    • 848e7a1 : Roll chromium_revision 7f27756..244de00 (366849:366862)
    • 2c56d7f : Roll chromium_revision 67b3d7f..7f27756 (366832:366849)
    • 6f94a45 : Roll chromium_revision 9beac25..67b3d7f (366816:366832)
    • b26f18d : Roll chromium_revision 1417e01..9beac25 (366759:366816)
    • f01f8c9 : Roll chromium_revision 2b7b555..1417e01 (366738:366759)
    • c12f7f4 : Roll chromium_revision 8a1fcdb..2b7b555 (366529:366738)
    • a6c86b2 : Revert "Enable IPv6 temporary address filtering on iOS."
    • 29488c2 : Enable IPv6 temporary address filtering on iOS.
    • 93c08b7 : Adding bit exactness test for Opus decoding in NetEq.
    • a72e734 : [rtp_rtcp] cleanup in RTCPSender class internals. PrepareReportBlock and AddReportBlock private functions merged: PrepareReportBlock moved report block from statistic to temporary structure AddReportBlock copied that temporary structure into temporary map right after. Thanks to rtcp packet classes that temporary structure is now unneccesary.
    • a8890a5 : rtcp::Nack packet moved into own file and got Parse function
    • 53c317c : Roll chromium_revision 4688e75..8a1fcdb (366364:366529)
    • cfb7f01 : Disable VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly due to flakiness on LinuxAsan.
    • e6bf587 : Deleted VideoCapturer::screencast_max_pixels, together with VideoChannel::GetScreencastMaxPixels and VideoChannel::GetScreencastFps.
    • db8cf50 : Fix two problems in network.cc: 1. It signals network changed events whenever there are more than one IP address in a network. 2. It does not signal network changed events if a network disconnects and connects again. Also changed DumpNetworks for better debugging.
    • 1227e8b : [rtp_rtcp] time helper functions RTP timestams helper functions moved from rtp_utility added functions to deal with CompactNtp timestamps
    • 5908c71 : Lint fix for webrtc/modules/video_coding PART 3!
    • f5b1abf : Roll chromium_revision c844be9..4688e75 (366322:366364)
    • de94c13 : Add webrtc/audio and webrtc/call to WATCHLISTS.
    • 9d3ab61 : Lint fix for webrtc/modules/video_coding PART 2!
    • ff48361 : Step 1 to prepare call_test.* for combined audio/video tests.
    • cce46fc : Lint fix for webrtc/modules/video_coding PART 1!
    • 5380532 : Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.
    • 9fca7e1 : A unittest that reports the statistics for the duration of an APM stream processing API call.
    • c693820 : CQ: Add linux_libfuzzer_rel trybot as default.
    • 54bab12 : Roll chromium_revision db567a8..c844be9 (366304:366322)
    • 2f042f2 : Roll chromium_revision 1b6c421..db567a8 (365999:366304)
    • a4df27b : Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
    • f4f5cb0 : Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
    • bd7d8f7 : Adding a MediaStream parameter to createSender.
    • 92594a3 : Moving FFT on farend signal to where it is used in AEC (bit exact).
    • 4ff818e : Make download_from_google_storage print less during runhooks.
    • 740c367 : iSAC: Remove unnecessary WEBRTC_LINUX define.
    • c155b16 : remove deprecated StringToIP() methods from SocketAddress API
    • 36d4c54 : Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ )
    • 455a252 : Fix pointer compare-and-swap on Windows.
    • c1cd566 : delete basictypes.h header
    • b7d9a97 : Expose codec implementation names in stats.
    • 6c6510a : audio_device: Move sources into platform-conditions.
    • 9b7fc7f : Defines for ARM and MIPS CPU types.
    • ae2c5ad : Added option to specify a maximum file size when recording an AEC dump.
    • 095ae15 : Keep listening if "accept" returns an invalid socket.
    • 88518a2 : Use NV21 instead of YUV12 and clean up.
    • 48477c1 : MediaCodecVideoEncoder, set timestamp on the encoder surface when drawing a texture.
    • fc96bd1 : Roll chromium_revision e78bc2f..1b6c421 (365856:365999)
    • 77fa59d : Fix build break in google3 import caused by https://codereview.webrtc.org/1532543003
    • 4638331 : DTLS-SRTP set up is bypassed when the channel has been writable.
    • efb047d : Compilation failed with openssl.
    • 933f3ec : Roll chromium_revision ddfc1fe..e78bc2f (365801:365856)
    • 002f0d0 : VP9: Set speed setting to 8 for ARM.
    • 5a4ce2f : Deleted declaration of VideoCaptureInput::DeliverI420Frame
    • a0b9549 : Roll gtest-parallel.
    • 369f828 : Adding trace events for the APM render and capture stream processing functions.
    • 9390f84 : Use std::nullptr_t instead of decltype(nullptr)
    • 1e0cfd9 : Add VP8 and H264 depacketizer fuzzers.
    • 9d98f21 : Roll chromium_revision 68898fb..ddfc1fe (365698:365801)
    • a689b44 : Add tracing to NetEqImpl::InsertPacket
    • 0eb15ed : Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
    • e376f0f : Add Windows Clang trybots to the default set.
    • e40dedb : Roll chromium_revision 004c7b4..68898fb (365580:365698)
    • a089257 : Cleanup use of "do { ... } while (0)".
    • a54a080 : Add ufrag to the ICE candidate signaling. On the receiving side, if a candidate arrives with an old ufrag, it will be dropped. If it contains a new frag that has never seen before, it will hold the ufrag and create connections, although those connections are not pingable until the ICE credentials are received. This could avoid a bunch of ICE generation issues.
    • 3514cbe : Add DrFuzz support to webrtc fuzzers.
    • 7cae30c : Disable warnings failing when using Clang on Windows.
    • 9f58795 : Roll chromium_revision 2c8eb1f..004c7b4 (365513:365580)
    • 361888c : OWNERS: Add * to .gyp{i,} everywhere.
    • 2f29d70 : Roll gtest-parallel.
    • 0bc176b : Further refactored the echo suppressor code: -Extended the InverseFft function to be more generally applicable. -Included the previous external extra scaling into the preexisting InverseFft call. -Moved the updating of aec-delayEstCtr to where it is actually used. -Refactored the output production and comfort noise addition using the InverseFft function. -Removed the if-statements checking the value of the constant flagHbandCn as any value different from 1 would crash the program. Also removed the constant
    • c482eb3 : Don't account for audio in the pacer budget.
    • 5f026d0 : Update NetEq network statistics in neteq_unittest.
    • 4430763 : AudioCodingModuleImpl: Stop failing artificially for non-Opus encoders
    • 99b1a32 : Retyped the frequency estimate of the comfort noise for the higher band to harmonize the AEC code. -Changed the type for the frequency estimate of the comfort noise for the higher band to be a two dimensional float array instead of a complex_t array. This makes sense since all the other frequency estimate (apart from the coherence) use this format and doing this change allows bundling the IFFT operations into using the InverseFFT method. -Moved the memset of the frequency estimate of the comfort noise to where it is used and made it conditional so that it is only performed when used. -Harmonized the if-statements for when the frequency estimate of the comfort noise is computed in the different optimized ComfortNoise computation methods.
    • 426ae9d : Roll chromium_revision 6e5b8cb..2c8eb1f (365419:365513)
    • a6db495 : Move Rent-A-Codec out of CodecManager
    • a29386c : Make VoiceDetection not a ProcessingComponent (bit exact).
    • 672aba3 : Fix error prone code in VideoCapturerAndroid
    • 66085be : Bugfix that fixes the error where the audio processing module is called using the wrong sample rate for the render signal.
    • 54999d4 : rtcp::Dlrr block moved into own file and got Parse function
    • 29e2f93 : Fix NoiseSuppression initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1507683006/.
    • 45fd9fe : New macro: RTC_DEPRECATED (for annotating deprecated functions)
    • ed644d8 : Roll chromium_revision bff4606..6e5b8cb (365226:365419)
    • eb45981 : Restoring behavior where PeerConnection tracks changes to MediaStreams.
    • 44f0819 : Fixing bug where "mid" wasn't preserved across re-offers.
    • c1316a1 : Fix HPF initialization behavior. This was changed when removing the ProcessingComponent inheritance in https://codereview.webrtc.org/1490333004/.
    • 95d9851 : Add speech encoder to the encoder stack specification struct
    • 7eb914d : Fix incorrect comment
    • 78315b9 : Reland of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1528043002/ )
    • f9945b2 : Only try to pair protocol matching candidates for creating connections. If the local port and the remote candidate's protocols do not match, do not even try to pair them. This avoids printing out confusing logs like "Attempt to change a remote candidate..." in p2ptransportchannel when two remote candidates have the same port number but different protocols.
    • 949028f : Make LevelEstimation not a ProcessingComponent.
    • 5e0218c : Revert of Base webrtc fuzzers on a template. (patchset #1 id:1 of https://codereview.webrtc.org/1524993002/ )
    • 5125433 : Android: Refactor renderers to allow apps to inject custom shaders
    • 91941ae : rtcp::VoipMetric block moved into own file and got Parse function
    • 32d989b : Disable transport sequence numbers for audio.
    • 10aea22 : Roll chromium_revision 53970fd..bff4606 (365141:365226)
    • 377b5e6 : enabled cpplint for the webrtc/modules/rtp_rtcp directory
    • 6eca7e3 : Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
    • 6db6cdc : [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
    • 9638143 : Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ )
    • e005cf2 : [rtp_rtcp] SSRCDatabase class cleaned (including all lint errors)
    • 5ea3da2 : Base webrtc fuzzers on a template.
    • 8f09f17 : Simple CL to fix lint errors in webrtc/modules/remote_bitrate_estimator. Added the lint check for the folder to the presubmit script.
    • 498ae00 : Disable ThreadTest.ThreeThreadsInvoke on DrMemory bots.
    • 47a740b : [rtp_rtcp] lint errors about rand() usage fixed.
    • 2d36b92 : Roll chromium_revision 10bf0e1..53970fd (365000:365141)
    • 1588793 : Fixing flaky LocalP2PTestSctpDataChannel test.
    • c9be007 : Fixing and re-enabling some flaky PeerConnection tests.
    • bd29246 : Reland of Free SCTP data channels asynchronously in PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1513143003/ )
    • 82ccfcf : Remove unused and rarely used LOG_ macros.
    • e22e1cb : Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ )
    • 40f349f : [rtp_rtcp] Lint errors cleared from rtp_rtcp/test
    • 3207916 : Made EglBase an abstract class and cleaned up. Adds EglBase10 that implemenents EglBase for EGL 1.0
    • 03960d9 : Roll chromium_revision 4bc4277..10bf0e1 (364953:365000)
    • bc14164 : Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ )
    • b2f80e3 : rtp_rtcp/test/BWEStandAlone deleted as obsolete
    • a78c021 : Add APK targets to build libjingle_peerconnection_unittests for Android.
    • 17821db : Wire up bandwidth limitation info to GetStats and adapt_reason.
    • ac921d7 : Add "x"s in the end of a stripped IPv6 address string.
    • 38bb8ad : Add test for verifying configured key frame interval for VP9.
    • e5ae6f8 : Correcting the check for the return code produced by AudioProcessing::ProcessReverseStream(). Before the change, only -1 was considered to be an error.
    • 1d5c19d : Address comments from code review 1505253004 (https://codereview.webrtc.org/1505253004/)
    • 4759bfb : Roll chromium_revision 7de03ed..4bc4277 (364770:364953)
    • aa32c3e : Update API for Objective-C RTCIceServer
    • cb95f54 : Remove pointless move() to fix build on clang/win.
    • 66679dc : Update WARN_UNUSED_RESULT macro to match Chromium's version.
    • be26c07 : Roll gtest-parallel.
    • b798f38 : Roll chromium_revision 710285b..7de03ed (364599:364770)
    • f888bb5 : Support for unmixed remote audio into tracks.
    • f67c548 : Handle Turn error response to RefreshRequest, CreatePermissionRequest, and ChanelBindRequest
    • 04e9146 : Discard old-generation candidates when ICE restarts The existing code only do so on the controlled side.
    • 43e4e23 : Remove thread-id wraparounds in event tracing.
    • 822bdf9 : Remove cricket::VideoEncoderConfig.
    • 4c1093b : Add FEC producer fuzzing and a unittest for one of the issues found.
    • 5b659c0 : Special-case android-arm64 in codec bitexactness tests
    • b562c33 : Remove ancient VoE suppressions.
    • cb23c0d : Adding Opus to RTPencode.
    • 71f5a9a : This cl change VideoCaptureAndroid to handle CVO the same way when capturing to texture as when using ordinary byte buffers.
    • 0b0a88b : Add aecdump support to AppRTCDemo
    • 4dfe332 : Roll chromium_revision 026b937..710285b (364421:364599)
    • 55bcf0f : Fix -Wformat error in Win-Clang build (take 2)
    • 013e83b : Fix -Wformat error in Win-Clang build
    • cf846ad : Adding stub files needed for https://codereview.webrtc.org/1507973003/
    • 7c73bdb : Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor.
    • ed83edc : Roll chromium_revision 2e451bf..026b937 (364330:364421)
    • 6a6f089 : in rtp_rtcp module: fixed build/namespaces lint errors fixed readability/namespace lint errors
    • a1f567a : Revert of Free SCTP data channels asynchronously in PeerConnection. (patchset #3 id:40001 of https://codereview.webrtc.org/1492383002/ )
    • 61a90f9 : clang/win: Fix -Wextra warnings in webrtc.
    • 5c1def8 : modules/rtp_rtcp/include folder cleared of lint warnings Functions that do not follow lint are marked deprecated, including function in the interface.
    • 796cfaf : Add VideoCodec::PreferDecodeLate The purpose is so that a decoder (Android) that only have a limited number of output buffers can make sure that decoding is done just before the frame is needed.
    • 4d68208 : Reduce the runtime of some ACM tests in modules_tests
    • c490e01 : Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to do the conversion using an opengl fragment shader.
    • b8b6fbb : lint build/include errors fixed in rtp_rtcp module
    • 90b9fc9 : Roll chromium_revision a02d286..2e451bf (364268:364330)
    • 866df66 : Typo fix: Enable a bunch of tests that were accidentally disabled
    • 5811a39 : Replace EventWrapper in video/, test/ and call/.
    • 0f2e939 : Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.
    • 162abd3 : lint whitespace warning removed from most rtp_rtcp/source/ files rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.
    • 84e78f9 : Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/.
    • 0b3d7ee : Prevent RTCP SR to be sent with bogus timestamp.
    • 48bf238 : Some further minor bitexact APM echo suppressor refactoring -Moved memsets to where their variables are used. -Removed redundant. -Changed a pointer scalar to be accessed in pointer notation rather than in array notation.
    • 5ba58c6 : Roll chromium_revision dad6346..a02d286 (363782:364268)
    • a6e4328 : Remove unnecessary test code on Windows.
    • 70625e5 : Enable cpplint for webrtc/examples and fix all uncovered cpplint errors.
    • 2e5fe31 : Remove myself from root_files watchlist.
    • 1387149 : Adding reduced size RTCP configuration down to the video stream level.
    • ee40821 : WebRTC: Update set of known root certificates
    • b14f001 : Some minor (bitexact) AEC echo suppressor refactoring -Moved filter reset from the echo suppression into the echo subtraction code where it belongs (the echo subtractor should own its filter reset). -Moved the selection between using the microphone sinal and the echo subtractor output down to the lowest level in the EchoSuppression function. This makes sense as that selection was very hidden in an unrelated sub-sub-function call and as the selection is critical for what the AEC outputs.
    • 434aca8 : Add empty placeholder files for remote audio tracks. This is needed for Chromium so that we can roll, update libjingle.gyp and then continue.
    • afeb438 : Moved code into the lowest level of EchoSuppression to simplify future refactoring and development.
    • d1590b2 : Lint clean video/ and add lint presubmit check.
    • 4cf61dd : NetEq: Add codec name and RTP timestamp rate to DecoderInfo
    • 3980d46 : RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime().
    • af3b9cb : Removing DrMemory suppresssion on PushResampler.
    • 5eb4988 : [rtp_rtcp] Lint build/header_guard errors fixed
    • 7623ce4 : Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
    • d3c9447 : Nuke TickTime::UseFakeClock.
    • bda7e0b : Fixing issue with default stream upon setting 2nd remote description.
    • d02b0fa : Add oldest rotation type option to RTCFileLogger
    • 5e465c3 : Make NoiseSuppression not a processing component (bit exact).
    • 1a9d615 : Add tracing to public PeerConnection methods.
    • 2d63680 : Roll chromium_revision 9dfb3a1..dad6346 (363718:363782)
    • 7b2f762 : Don't call SetPreviewFormat if capturing to textures. This fix an issue seen on Huawei Y300 where the camera feed is black and white if we capture to textures and setpreviewformat is called.
    • edd8fef : Add new view that renders local video using AVCaptureLayerPreview.
    • 70f9903 : Make HighPassFilter not a ProcessingComponent anymore (bit exact).
    • 246b817 : Refactor handling of AudioOptions.
    • 8237abf : Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
    • e10c82d : Deletes temporary files that are generated in several ACM unittests.
    • d7b7ae8 : Add encode/decode time tracing to audio_coding.
    • 9f45a45 : Add tracing to upper-level WebRTC calls.
    • cd6f539 : Revert of RTCCertificate::Expires() and ::HasExpired() implemented (patchset #5 id:140001 of https://codereview.webrtc.org/1494103003/ )
    • fe32a76 : Create fuzzer tests for audio decoders
    • ffea13c : PRESUBMIT: change native API check from warning to information.
    • 20ef654 : RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime().
    • 325b345 : There was an old scaling for CNG 48 kHz in the code, from the time where Audio Coding Module didn't have full 48 kHz support. This CL removes the scaling.
    • 88eeac4 : Adding video_processing to presubmit lint check
    • 4654d20 : Add test which verifies that the RTP header extensions are set correctly for FEC packets.
    • 99ab944 : Clang format of video_processing folder.
    • a440c6f : Roll chromium_revision 3b8be21..9dfb3a1 (363445:363718)
    • 3868692 : Free SCTP data channels asynchronously in PeerConnection.
    • 46ad542 : Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
    • 6f28cf0 : Implement standalone event tracing in AppRTCDemo.
    • 84f0970 : Reland of "Create rtc::AtomicInt POD struct."
    • 0f490a5 : Add logs when stun or turn host lookup is completed. This will help investigate issues caused by DNS lookup.
    • cd4003f : Use @webrtc.org addresses for OWNERS.
    • cf890bc : Roll gtest-parallel.
    • 0608dc5 : Roll chromium_revision 4918765..3b8be21 (363393:363445)
    • 5f6deaf : Remove unused RTP-header parser.
    • ab82cbb : Disable RtcEventLogTest.DropOldEvents on memcheck.
    • 03671cb : Use existing parser in ReceivesAndRetransmitsNack.
    • fc47ed6 : rtcp::Rrtr block moved into own file and got Parse function
    • 1aa420b : Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead.
    • 9d69c3f : Return a copy of the supported RTP header extensions instead of a reference.
    • b86d4e4 : Prepare the AudioSendStream to be hooked up to send-side BWE.
    • 03f80eb : Refactor EglBase configuration.
    • a856542 : Initial VideoProcessing refactoring.
    • 2512f44 : Roll chromium_revision 292ab9f..4918765 (363376:363393)
    • c9f1cb8 : Roll chromium_revision 72c3265..292ab9f (363365:363376)
    • 34a7054 : Roll chromium_revision 626eecf..72c3265 (363027:363365)
    • 1218d7a : Allow remote fingerprint update during a call
    • 86aaa4b : Revert "Allow remote fingerprint update during a call"
    • 9c38c2d : Allow remote fingerprint update during a call
    • 381b421 : Ping backup connection at a slower rate and make it configurable from the app. Changed the decision on whether a connection is pingable: 1.Check whether a connection is a backup connection. A connection is considered as a backup connection only if the channel is complete, the connection is active and it is not the best connection. 2. Ping a non-backup connection if it is active and for backup connection, ping it at a slower rate. Note the default behavior is the same as before.
    • 45b0efd : Stop sending stun binding requests after certain amount of time.
    • 9e1b992 : Clear old decoders after recreating the receiver.
    • 97f7e13 : rtcp::ReceiverReport moved into own file and got Parse function
    • 7c704b8 : Use webrtc/base/logging.h in stefan@'s ownership.
    • b572768 : - Remove calls to VoEDtmf from WVoE/MC. - Flatten logic and make the relevant calls on VoE::Channel from AudioSendStream::SendTelephoneEvent(). - Store current payload type for telephone events in WVoMC, instead of setting it on the Channel. This should be refactored to be an AudioSendStream::Config parameter when we redo WVoMC::SetSendCodecs().
    • fcdcf4a : Disable RtcEventLogTest.DropOldEvents on DrMemory.
    • 66f7f4e : Roll chromium_revision d3aa9b1..626eecf (362950:363027)
    • fd59523 : Add webrtc/base to deprecated APIs.
    • bc32ab4 : Remove 'video_engine_core_unittests' binary.
    • ff24c04 : Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations.
    • 1a5cf6e : Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine).
    • f7c5776 : Refactorings to send RTCP packets directly via the RtcpPacket callback, with some simplifications enabled by this. NACK now also sent via RtcpPacket.
    • 9cf0c3d : Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient.
    • d048aa0 : Make the audio codecs' GN targets self-sufficient
    • b4a1ae5 : Add separate send-side UMA stats for screenshare and video.
    • 29e3003 : Bring back baremetal trybots to the default set.
    • 5385554 : Roll chromium_revision 7461ceb..d3aa9b1 (362933:362950)
    • a4527c8 : Add comments about the Audio parts of the public Call API being WIP.
    • 74a5ffb : Roll chromium_revision f068d2f..7461ceb (362762:362933)
    • 631e134 : Rewrote the thread synchronization parts of the test for the locking in APM in response to a locking problem when running in a single-threaded manner.
    • 917ba52 : autoroll: Update Clang script path.
    • 53047c9 : Add PRESUBMIT check for native API changes.
    • c3e0fe7 : Make it extra safe when deleting a turn entry.
    • 7635684 : Fix Mac ObjC PeerConnection API compilation.
    • 9462052 : In some rare Android systems ConnectivityManager may be null. Handle this case more gracefully.
    • a448607 : Roll chromium_revision a45c85a..f068d2f (362609:362762)
    • 3c28d0d : Disable PeerConnectionEndToEndTest.Call on Mac.
    • 1d63dd0 : - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. - Remove the DF_PLAY/DF_SEND flags, only allow sending.
    • ee524f7 : Adding Java binding for CreateSender.
    • de0fc58 : Adding two more debug macros for logging scalar values to file.
    • 7e4e01a : Add header extension filtering for WebRtcVoiceEngine/MediaChannel.
    • 2515af2 : Removing some unnecessary string manipulation code from VoEBase::GetVersion().
    • d20d247 : Fix VideoCaptureAndroid, drop frame when switching camera using textures. Dropping the first frame intended to fix a problem when switching cameras on N6 when we are capturing to textures but due to a silly bug fixed in this cl the frame was not dropped...
    • 226a602 : Fix problem when drawing to the Android Media encoder surface. Problem seen on N6. BUG=webrtc:5147
    • c729032 : Resolves issue with multiple calls to audio unit initialization
    • 40455d6 : This cl change so that we use EGL14 where it is supported and EGL10 otherwise. The idea is to make this agnostic to an application and for WebRTC except in EGLBase.
    • e338499 : Revert of Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations. (patchset #18 id:580001 of https://codereview.webrtc.org/1437463002/ )
    • 43b4806 : Add tests for vp9 (non-flexible mode) using different spatial and temporal configurations.
    • 06104b8 : Roll chromium_revision eeff895..a45c85a (362465:362609)
    • 41b0798 : Adding CreatePeerConnection method that uses new PC Initialize method.
    • 62a91ee : Roll chromium_revision 35f35af..eeff895 (362385:362465)
    • 187db63 : Remove VideoReceiveStream deregister of decoders.
    • 04a6bb9 : Roll chromium_revision f9fedae..35f35af (362322:362385)
    • f94abf7 : Nuke webrtc/common_video/plane.*.
    • dfbb3a4 : Simplify CodecManager::RegisterEncoder()
    • 46c9cc0 : Provide method for returning certificate expiration time stamp.
    • ea07373 : Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.
    • 0de97f1 : WebRtcVideoCapturer: SetCaptureState(CS_STOPPED) on Stop and ensure state changes in unittest.
    • ec192bd : Revert of Add _decoder CHECK to VCMGenericDecoder constructor. (patchset #2 id:20001 of https://codereview.webrtc.org/1485713002/ )
    • cb9792e : Fix VideoCapturerAndroidTest.testStartWhileCameraIsAlreadyOpen on Android M.
    • 9f8d39d : Add simple end to end test for video capture and encode using textures.
    • 021282f : Roll chromium_revision 47ce5fe..f9fedae (362117:362322)
    • 14f4144 : Add helper KeepRefUntilDone. The callback keeps a reference to an object until the callback goes out of scope.
    • ee69ed5 : Add separate event for camera freeze.
    • 70c0e29 : Disable PeerConnectionEndToEndTest.Call for TSan.
    • f893df3 : Add third_party/libc++static to .gitignore
    • a443ec1 : Add _decoder CHECK to VCMGenericDecoder constructor.
    • 7640ffa : Initialize type_preference_ in TestPort.
    • f9203c6 : Roll chromium_revision faa24ae..47ce5fe (362083:362117)
    • 99f8566 : Roll chromium_revision 0da9346..faa24ae (362069:362083)
    • bdf001a : Roll chromium_revision 8f57310..0da9346 (362067:362069)
    • 90728b9 : Roll chromium_revision 3e15d1a..8f57310 (362064:362067)
    • 4c14254 : Roll chromium_revision df4d569..3e15d1a (362055:362064)
    • df3efa8 : Introduced the new locking scheme
    • 3236b91 : Roll chromium_revision c54812d..df4d569 (362052:362055)
    • 535727e : Roll chromium_revision 5ac8f02..c54812d (362046:362052)
    • ae54b83 : Android SurfaceViewRenderer: Add resetStatistics() method
    • 43f1809 : Roll chromium_revision 7b99051..5ac8f02 (361977:362046)
    • 2fe1cb0 : Don't overwrite audio stats when they're not available.
    • 7e43138 : -Removed the state as an input to the FilterAdaptation function. -Renamed the TimeToFrequency and FrequencyToTime functions. -Moved the windowing from the TimeToFrequency function. -Simplified the EchoSubtraction function.
    • 19822d6 : audio_coding: Cleanup duplicated headers after "main" removal.
    • 358057b : Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.
    • ad85622 : Use webrtc/base/logging.h for voice_engine.
    • def5820 : Default to LS_INFO logging for release builds.
    • 521af4e : Remove duplicate decoders in BitrateEstimatorTest.
    • 395c7c6 : Re-add missing return in RegisterExternalDecoder.
    • f8385ad : rtcp::Pli moved into own file and got a Parse function Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.
    • e997a7d : Call InitDecode with proper resolution.
    • 795dbe4 : Remove RegisterExternal{De,En}coder error codes.
    • 34873b5 : Roll chromium_revision 7ec1eb8..7b99051 (361868:361977)
    • 26c8c91 : Using Rent-A-Codec for static Codec access in WVoE/MC.
    • 8779a77 : Fix standalone denoiser Android GN compile failure
    • 81b9bfe : Added a threadchecking scheme to APM that checks that the APM API calls are called from the correct threads. The actual threadcheckers were, however, removed and will be reintroduced in another upcoming CL.
    • 64c0a0a : Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ )
    • 42f580e : Leaving all original files in talk/app/webrtc/objc until we can officially tell clients about the new locations.
    • b1ac203 : Introduce helper class NtpTime Seconds and fractions parts of the ntp time presented with two values, but used as one. This helper structure can make that use more clear. (initially introduced into rtp_rtcp as https://codereview.webrtc.org/1435833003)
    • 6e40c09 : Fix root_files WATCHLIST.
    • 8c38e8b : Clean up PlatformThread.
    • ad113e5 : Fix bug in calculation of averge queue time in paced sender.
    • 226befe : Rewrote pacer and bandwidth UMA stats.
    • 06e05a8 : Make overuse estimator one dimensional.
    • 0fcaf99 : Enable cpplint for webrtc/video_engine
    • 727dbc2 : VideoCapturerAndroid - allow lower frame rate in bad lightning Insted of using a fixed frame rate, we allow the camera to use a lower frame rate. The camera will choose depending on lightning condition.
    • 871c419 : Add fuzzing of VP8 QP parsing.
    • c5b4c9b : Roll chromium_revision 664fe1e..7ec1eb8 (361806:361868)
    • 598242a : Support texture scaling in Androids MediaEncoder. This cl make it possible for the hw video encoder to downscale a texture image before encoding. The purpose is to allow downscaling if the quality is too bad at the current resolution. BUG=webrtc:4993 R=magjed@webrtc.org, pbos@webrtc.org
    • 3e6db23 : audio_coding: remove "main" directory
    • a3c20bb : Add support for scaling textures in AndroidVideoCapturer. The idea is to also reuse AndroidTextureBuffer::CropAndScale when scaling in the encoder.
    • fd5dae3 : Build/use constructormagic.h unconditionally.
    • 8f9902a : Standalone denoiser (off by default).
    • 96cb530 : Removed api call that will break the upcoming thread checking scheme
    • c03bdf9 : Roll chromium_revision aa8e58a..664fe1e (361601:361806)
    • 26ab91b : Add symlink to src/third_party/libc++-static
    • cdb38e5 : Strip IP addresses in NDEBUG (release) builds.
    • b86c502 : Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
    • a34c39e : GetDefaultLocalAddress should return false when the address is invalid
    • 89d658f : Fix fuzzer breakage in Chromium.
    • 11e0229 : Move Chromium logging into rtc_base_approved.
    • 6e004a4 : Revert of Created a test that reports the statistics for the duration of APM stream processing API calls. (patchset #15 id:280001 of https://codereview.webrtc.org/1436553004/ )
    • 675d437 : WIP: Changes after merge commit 'cb3f9bd'
    • fac0655 : Reland of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
    • 376e123 : Destroy a Connection if a CreatePermission request fails.
    • 1372508 : Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.
    • 54eb5e2 : Removed the aec state as an input parameter to the FilterFar function.
    • 880896a : A unittest that reports the statistics for the duration of an APM stream processing API call.
    • 9cd5c8c : Move the FEC enabling logic from CodecManager to Rent-A-Codec
    • 989b4ab : Move the stereo-disables-CNG logic from CodecManager to Rent-A-Codec
    • 46a491b : Set mac_deployment_target default to 10.7
    • 65ef319 : Roll chromium_revision b1d79c3..68cf0b8 (361146:361406)
    • 444682a : Remove frame time scheduing in IncomingVideoStream
    • 953eabc : Revert "GetDefaultLocalAddress should return false when the address is invalid"
    • 67c6df6 : GetDefaultLocalAddress should return false when the address is invalid
    • 7d842d6 : Move thread_ conditional back under defines.
    • c661213 : Skip setting thread priorities in NaCl.
    • b251472 : Add JNI interface for functions to start and stop recording AEC dumps and RTC event logs.
    • c3ddb3e : Improve documentation for ArrayView
    • 4c5eea3 : Android SurfaceViewRenderer: Don't rely on widthSpec/heightSpec after onMeasure() returns
    • b7a8829 : Remove duplicated headers after updating downstream code.
    • 302c978 : Work around data race in TransmitMixer.
    • 7baf79f : Temporary remove spamming MediaDecoder log This log will write for each decoded frame if the textures are rendered using VideoRenderGUI and the the screen is locked.
    • 92f8dbd : Remove VIDEOCODEC_* from engine_configurations.h.
    • 4f2152e : Android SurfaceViewRenderer: Make sure not to call eglCreateSurface() twice
    • 97c821d : Inline ConvertToSystemPriority.
    • 9237559 : Add SurfaceTextureHelper.disconnect(Handler handler) method This method should be used when the SurfaceTextureHelper is created to use a specific handler. This now guarantee that the looper used by handler is destroyed after a frame has been returned.
    • d480153 : Add option to capture to texture in AppRTCDemo for Android.
    • 978244e : Adding a bunch of Agora IO team members to the watch lists
    • d860523 : First part of the preparatory work before the actual work for solving the ducking problem starts.
    • 70bed7d : GN: Fix iOS error in audio_device and rtc_base
    • b5cb19b : Fixing direction attribute in answer for non-RTP protocols.
    • 05816eb : Fix target_arch for ios devices
    • 12411ef : Move ThreadWrapper to ProcessThread in base.
    • 255d6f6 : Test case for CL 1437933002.
    • 9c80bbe : Roll chromium_revision e038f1d..b1d79c3 (361088:361146)
    • 1aa6efe : Android ThreadUtils: Make the class public for access outside org.webrtc
    • 057fb89 : Add new method AcmReceiver::last_packet_sample_rate_hz()
    • 74e35f1 : Remove the special case for std::vector in rtc::ArrayView
    • d89814b : NetEq: Add new method last_output_sample_rate_hz
    • dfafd12 : Remove ThreadWrapper::GetThreadId. The method just calls rtc::CurrentThreadId(), which also has a more descriptive name.
    • 62e9bda : Implement fuzzing of VP9 depacketization.
    • ee37de3 : Add screenshare perf tests with lossy links
    • 1379f1f : Extract the parameters for the encoder stack from the CodecManager
    • 30a5e56 : Roll chromium_revision 3796a7a..e038f1d (361065:361088)
    • f0a431a : Exclude EndToEndTest.ReceivesTransportFeedback and TransportFeedbackNotConfigured from DrMemory.
    • db81ffd : Request keyframe if too many packets are missing and NACK is disabled.
    • fa8ae9a : Remove iostream include from file_audio_device.cc
    • 8becec3 : talk: remove deprecated *processor.h files
    • 87d5845 : Fix androidmediadecoder_jni TS logging. And fix pragma warning about deprecated "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h include.
    • c3c4cdb : Add Android x86 and x64 trybots to CQ.
    • d5674c3 : Roll chromium_revision c29f20c..3796a7a (361043:361065)
    • 50c5136 : RTCP Bye packet moved to own file Bye class got support for Parsing Reason field implemented
    • c982913 : Roll chromium_revision 6018759..c29f20c (361030:361043)
    • 485b5b0 : Roll chromium_revision 4df2d47..6018759 (361029:361030)
    • 82581a0 : Roll chromium_revision 3966d2c..4df2d47 (361020:361029)
    • b4a29d9 : Roll chromium_revision b854092..3966d2c (360794:361020)
    • 13f6b8f : Increase transport feedback frequency to 20 Hz.
    • 43edf0f : Require negotiation to send transport cc feedback over RTCP.
    • bd13838 : Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.
    • 672304a : NetEq: Remove overly verbose logging
    • 5def7b9 : Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
    • 7add058 : Move some receive stream configuration into webrtc::AudioReceiveStream.
    • 6834fa1 : Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
    • 0a43fef : Allow pacer to boost bitrate in order to meet time constraints.
    • 34911ad : Improved error handling in iOS ADM to avoid race during init
    • 76a31ca : Avoids hitting DCHECK in OpenSL ES player
    • 1afae74 : Roll chromium_revision 5c83f4e..b854092 (360728:360794)
    • 30e9182 : This cl add support to encode from textures to MediaCodecVideoEncoder.
    • 5663b4f : iOS: Set enable_protobuf=1 by default.
    • 7e63ef0 : Allow default audio receive channel to receive on any unsignalled SSRC.
    • b0ad43b : Add aecdump support to audioproc_f
    • ceb450b : Roll chromium_revision c8eec9a..5c83f4e (360565:360728)
    • 17c0aff : Enable VP9 HW decoder on Exynos chips.
    • 7593aad : Re-enable mistakenly disabled PEM tests. Misc cleanup and alignment fixes.
    • 7755e20 : Chrome has now been updated.
    • 726b1f7 : Removed dummy "mediastreamsignaling.h"
    • 191c1f9 : Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots.
    • 12e21a0 : Remove dead code (we no longer support SILK)
    • ef45323 : Android: Make classes non-final
    • 062e14e : Roll chromium_revision bb7899a..c8eec9a (360504:360565)
    • f399f21 : Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on linux due to flakiness on the Linux64 Debug bot.
    • f22695c : Remove build_with_libjingle and exclude failing iOS tests from 'All' target.
    • 1503867 : Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots.
    • e488a0d : Fix DTLS packet boundary handling in SSLStreamAdapterTests.
    • 87097a8 : Roll chromium_revision ed2e3fb..bb7899a (360379:360504)
    • b6755ab : Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ )
    • 488e75f : Patchset 1 yet again relands without modification https://codereview.webrtc.org/1422963003/ It do the following:
    • 0969398 : Revert of Remove android_rel from CQ since all of its machines are offline. (patchset #1 id:1 of https://codereview.webrtc.org/1459083002/ )
    • 392d0c2 : Remove android_rel from CQ since all of its machines are offline.
    • 521ed7b : Reland Convert internal representation of Srtp cryptos from string to int
    • 318166b : Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
    • 2764e10 : Convert internal representation of Srtp cryptos from string to int.
    • 3c652b6 : modules/audio_coding: Remove some codec include dirs
    • b7ce964 : modules/video_coding/utility: Remove include
    • 1b20d81 : Roll chromium_revision 64f2817..ed2e3fb (360275:360379)
    • 0f59a88 : modules/video_processing: refactor interface-include + more.
    • ed7d6ec : WebRTC: Add compability header for video_coding refactoring.
    • ad948c4 : Preliminary support of VP9 HW encoder on Android.
    • 2557b86 : modules/video_coding refactorings
    • 4dd7a65 : Temporarily disable VERIFY while bug is investigated.
    • 223692a : Remove dead code
    • e1a27d4 : Move CNG/RED payload type extraction to Rent-A-Codec
    • 49a6c99 : Disables BitrateEstimatorTest.SwitchesToASTThenBackToTOFForVideo on win_drmemory_full due to flakiness.
    • 2446e5a : Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation
    • 0219c9b : rtcp::App moved into own file and got Parse function
    • 2aff615 : Remove spammy logging of RTCP delivery failures.
    • f70568c : So long and thanks for all the code reviews!
    • cb50c96 : Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently).
    • aa45843 : Roll chromium_revision a6d9f7f..64f2817 (360123:360275)
    • 310b093 : Fix active tcp port to 9
    • 2935e01 : Several Tick counter improvements try #2."
    • c073615 : Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc.
    • 0a75749 : Roll chromium_revision 04756fa..a6d9f7f (360053:360123)
    • 32f3996 : Re-apply change https://codereview.webrtc.org/1426673007/
    • 5c489c9 : Add OpenSL ES enable setting to AppRTCDemo (part 2).
    • 2be7c54 : Remove ViEEncoder::ScaleInputImage.
    • bd05f0b : Unconditionally build VP9 support.
    • 18adf0a : Add UMA for send bwe and pacer bitrate.
    • d9eec76 : Trace encoding/decoding time in a generic way.
    • 5a71f03 : Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant since it is recommended for VoIP apps.
    • 45e998d : Roll chromium_revision a2e8a40..04756fa (359987:360053)
    • fd614c2 : Adding thread timeout for audio recorer thread in Java
    • e663392 : Add OpenSL ES enable setting to AppRTCDemo.
    • 3c12f4d : Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ )
    • 192164e : Preparational work before introducing the locks in order to harmonize the code: -Moved the initialize function -Moved api_format into the shared state
    • 4d291f7 : Applied the render queueing to the agc.
    • 03179cd : Roll chromium_revision 6fd4bdd..a2e8a40 (359891:359987)
    • 740c4f1 : Remove packet initializer in RtpRtcpRtxNackTest.
    • 854e84c : Use webrtc/base/logging.h for video coding/processing.
    • c91d173 : Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ )
    • fa6228e : Introduced the render sample queue for the aec and aecm.
    • 4c27e4b : Several Tick counter improvements.
    • eb8b388 : Fix VP9 support in AppRTCDemo.
    • 6f8ce06 : common_video: rename interface - include
    • 591cb1f : Roll chromium_revision c958aa7..6fd4bdd (359816:359891)
    • b27f590 : Create rtc::AtomicInt POD struct.
    • 3528a27 : Flesh out webrtc/.gitignore
    • 482b12e : Remove BundleFilter filtering of RTCP.
    • 8b85de2 : Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail.
    • 9a7c838 : Adding stddef.h to opus_inst.h.
    • 3a94154 : Move some send stream configuration into webrtc::AudioSendStream.
    • 633a3aa : ThreadUtils: Add joinUninterruptibly() with timeout
    • e155ae6 : Move CNG and RED management into the Rent-A-Codec
    • 54e9232 : Revert of Do not delete the turn port entry right away when the respective connection is deleted. (patchset #5 id:260001 of https://codereview.webrtc.org/1426673007/ )
    • 2a654fa : Roll chromium_revision cad2987..c958aa7 (359796:359816)
    • 0b9e29c : Remove include dirs from modules/{media_file,pacing}
    • 3e0f602 : Android EglBase: Add support for creating EGLSurface from Surface, not SurfaceHolder
    • d9b75be : Fix a data race in the thread unit tests.
    • 6f14be8 : Add limit for minimum number of required samples before recording input and sent framerate stats.
    • 3c735f4 : Roll chromium_revision b77e5bb..cad2987 (359767:359796)
    • 8c64860 : Roll chromium_revision 3b7968d..b77e5bb (359482:359767)
    • e58fe8e : Do not delete the turn port entry right away when the respective connection is deleted. BUG=webrtc:5120
    • fa5d0db : cleanup: get rid of basicdefs.h include
    • a4845ef : Fix flaky tests
    • 4a41361 : Android SurfaceViewRenderer: Never hold a pending frame indefinitely
    • c01c254 : Revert of Android MediaCodecVideoDecoder: Manage lifetime of texture frames (patchset #12 id:320001 of https://codereview.webrtc.org/1422963003/ )
    • f8506cb : rtcp::Ij renamed to rtcp::ExtendedJitterReport to match name given in the RFC5450 private member renamed to inter_arrival_jitters_ for the same reason. rtcp::ExtendedJitterReport moved into own file accessors and Parse function added to make class usable for parsing packet
    • cbe9f51 : Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ )
    • 0fa9b22 : Remove scoped_ptrs for VCM sender_ and receiver_.
    • df948f0 : rtcp::ReportBlock refactored to contain parsing
    • 0a41893 : Remove BitrateController dependency fromVideoReceiveStream.
    • 464c087 : Rename screenshare test.
    • 0e7e259 : Move BitrateAllocator from BitrateController logic to Call.
    • 69191ed : Roll chromium_revision 4771dd5..3b7968d (359351:359482)
    • faac497 : Fix for scenario where m-line is revived after being set to port 0.
    • 69d0d46 : Roll chromium_revision e658ee0..4771dd5 (359300:359351)
    • 2cd7afe : Do not delete a connection until it has not received anything for 30 seconds.
    • 8597543 : Schedule a CreatePermissionRequest after the success of a previous request unless a channel binding request is already scheduled.
    • 68876f9 : Introduces Android API level linting, fixes all current API lint errors.
    • 56a34df : Re-add a thread check in Call::Call that was removed by mistake in a rebase.
    • 9576e54 : Reland "Prepare MediaCodecVideoEncoder for surface textures.""
    • 8093d54 : Change default SSRC for RTCP receiver reports to not collide with video.
    • dfe434e : Roll chromium_revision b0415d9..e658ee0 (359214:359300)
    • 5dda80a : Remove webrtc/modules/video_{capture,render}/include
    • e71b24e : OpenSL ES stability improvements.
    • fc6affc : Android SurfaceViewRenderer: Call glClear() for every frame to avoid bad GL state
    • 9683964 : Trivial initialization fix in AudioDeviceIOS
    • 31c8167 : Roll chromium_revision 7e059f9..b0415d9 (359143:359214)
    • a8e9f5e : A little cleanup in p2ptransportchannel and transportchannel. No functional changes.
    • 066ded9 : Relax the stun ping check on valid result.
    • 33daa7e : Roll chromium_revision 4a38519..7e059f9 (359080:359143)
    • 6b14f93 : Adjust parameter for VP9 resize unittest.
    • 9b5ee9c : Send back ping response if the ping comes from an unknown address. BUG=webrtc:5171
    • 653b8e0 : Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ )
    • 9b72af9 : Remove webrtc/modules/audio_processing/{aec,aecm,ns}/include
    • e03cab9 : When running this code in chromium on a machine with IPv6 disabled, the RTC_DCHECK fails and in release build, it could leak to further crash in chromium's rtc_peer_connection_hanlder.cc.
    • ee2bac2 : AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments
    • 91d9260 : Add receive bitrate UMA stats.
    • 4dc9411 : CodecManager::RegisterEncoder: Call SetFec on new encoder, not old
    • 718b6c7 : Add waiting to SetSendSsrc tests.
    • 4b56904 : Fix race in VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent.
    • 00ac85e : Update temporal up switch field for non-flexible mode according to updates in the RTP payload profile. The U bit is no longer obtained from the SS data.
    • f616a35 : Roll chromium_revision 5a2ae99..4a38519 (359027:359080)
    • fa566d6 : Remove webrtc/examples/android/media_demo.
    • cbfabbf : Fix potential tearing issue in VideoRendererGui. This make sure that the texture copy is syncronized.
    • 9cb8982 : Patchset 1 is a pure revert of "Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/
    • b2d1c50 : SurfaceViewRenderer: Add resource name to log outputs and exceptions
    • 1323fc3 : Remove webrtc/test/channel_transport/include
    • 5237aaf : Convert usage of ARRAY_SIZE to arraysize.
    • e134a53 : Roll chromium_revision 6f156f7..5a2ae99 (358880:359027)
    • ad13d2f : Round Rate computations from RateTracker.
    • 9cafd97 : Remove global list of SRTP sessions. Instead save a reference to the SrtpSession inside the srtp_ctx_t.
    • 9af97f8 : WebRTC should generate default private address even when adapter enumeration is disabled.
    • 542059e : Roll chromium_revision bff0bbb..6f156f7 (358822:358880)
    • be57983 : Rename Maybe to Optional
    • 5376100 : Add icu as a dependency on Android.
    • 69a7fd5 : Support VP9 HW video decoding on Android.
    • ed8275a : CodecManager: Eliminate the stereo_send_ member
    • a34bb2a : Remove icu as a dependency
    • c94bd9b : If a desktop captured window switches on/off it full screen mode, the capture may be unexpectedly terminated. During the transition of full screen mode on/off, the window can be temporarily invisible.
    • d153a37 : Remove contention between RTCP packets and encoding.
    • cfc319b : Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ )
    • c95c366 : Move the Rent-A-Codec™ from CodecOwner to CodecManager
    • cf3e13d : Roll chromium_revision 95473df..bff0bbb (358772:358822)
    • 0be8f1d : Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
    • 3ed3487 : Remove field trial check for VP9. VP9 is put as second codec in supported codec list.
    • 327d8ba : Add DecodedImageCallback::Decoded() function with custom decode time value. On Android, we would like to use MediaCodec output buffers to hold decoded frames until they can be rendered to a texture. There can only be one texture buffer used at the same time and therefore the calculated decode time in VCMTiming will be wrong since that calculation will also include the time where the decoder waited for the upper layers (that depend on network jitter and actual render time) to release the frame.
    • 805fc71 : Let Rent-A-Codec™ create and own speech encoders
    • 3cea256 : Reland "Prevent Opus DTX from generating intermittent noise during silence"
    • 626252f : Adding minyue@ to some watch lists.
    • 77ccfb4 : Work on flexible mode and screen sharing.
    • ce83ae1 : Improve informative message in codereview.settings.
    • c12be39 : -Removed the indirect error message reporting in aec and aecm. -Made the component error messages generic to be an unspecified error message.
    • 952892a : Fix a 64-bit pointer truncation bug found by VC++ 2015
    • e36de90 : Roll chromium_revision 7818e07..95473df (358577:358772)
    • b4a753f : Revert of Prevent Opus DTX from generating intermittent noise during silence (patchset #10 id:250001 of https://codereview.webrtc.org/1415173005/ )
    • c1cd2bb : Turned off progress report for finished processing when the progress report is explicitly deactivated
    • f475add : Prevent Opus DTX from generating intermittent noise during silence.
    • ab48ef3 : Remove legacy audio device glue files.
    • 83dfad6 : VideoCapturerAndroid: Changed camera freeze check to check that all frames are pending before reporting a client error.
    • 48407f7 : Changed queue implementation to the proposed vector-based solution. Added unit tests.
    • cbe0cd4 : Roll chromium_revision 5477ec0..7818e07 (358562:358577)
    • 1f1912d : Added unittest of the locking functionality in the audio processing module The test is currently disabled as it takes too long to run in a coffe-cup manner
    • 89ef6cc : Attempt to open Android camera later if it is already in use.
    • 1ebf8ba : SurfaceViewRenderer: Drop old frames instead of new frames
    • e15303b : Roll chromium_revision 78fd1c0..5477ec0 (358554:358562)
    • b17900c : Roll chromium_revision 56d4322..78fd1c0 (358550:358554)
    • 3bfef44 : Changed timeout to 6s for reporting android camera freeze. Also distinguish between camera failures and failures due to that buffers has not been returned. Adds unit tests for making sure CameraEventHandler.onError is triggered if frames are not returned.
    • e2e1de4 : Roll chromium_revision fa2f634..56d4322 (358544:358550)
    • b8c5af7 : Roll chromium_revision 4598ddc..fa2f634 (358535:358544)
    • 0ad924f : Roll chromium_revision e99453b..4598ddc (358523:358535)
    • 110c19b : Roll chromium_revision a469250..e99453b (358505:358523)
    • a40f2ec : Roll chromium_revision e2bf349..a469250 (358372:358505)
    • 39d8bee : Make ACMCodecDB private to RentACodec
    • 566ef24 : Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
    • 19299fb : Remove interface directories kept to avoid breaking downstream.
    • d6c0f8c : Remove ACMCodecDB::Codec, and make the rest of ACMCodecDB private
    • 23725e0 : Remove ICU usage from jni_helpers.cc.
    • a821afe : Roll chromium_revision 736929d..e2bf349 (358303:358372)
    • bb79127 : Add Riku Voipio to AUTHORS.
    • d66daa2 : Removed cname and receiver_reference_time_report from proto and logging code. Changed logging of RTCP to omit messages of type SDES and APP.
    • 56b1128 : Change to use local Random object instead of global rand() in the RtcEventLog unit test.
    • 698aa8d : Cleanup workaround for grit version dependency
    • c4a1c37 : Removed vie_defines.h
    • d812e14 : Roll chromium_revision d3da94c..736929d (358098:358303)
    • dc0da59 : Remove old system_wrappers event_tracer.h.
    • fb3d8b3 : Remove ACMCodecDB::CodecFreq
    • 288886b : Pass audio to AudioEncoder::Encode() in an ArrayView
    • 1a4e9d7 : Roll chromium_revision 74a3f59..d3da94c (358040:358098)
    • c253a1c : Reland of "Change type of pid_diff (int16_t - uint8_t) according to updates in RTP payload profile."
    • b7a5c16 : Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
    • 006d93d : Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estimator.
    • 962c5ce : Re-enable VP9 resize test.
    • cc4989a : Remove BaseUnitTest suppression.
    • 586c066 : Remove DrMemory gtest_exclude for dead test.
    • b0a078b : Roll chromium_revision b483788..74a3f59 (358023:358040)
    • 4de30ac : Update moved find_depot_tools.py script location
    • 93a2feb : Move ACMCodecDB::ValidPayloadType to RentACodec
    • 86b4050 : Reland of Add aecdump support to audioproc_f. (patchset #2 id:250001 of https://codereview.webrtc.org/1423693008/ )
    • d279941 : Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
    • 394c537 : Update layer indices for non-flexible mode according to updates in the RTP payload profile.
    • f97bfed : Revert of Move audioproc_test_utils into enable_protobuf condition. (patchset #1 id:1 of https://codereview.webrtc.org/1419533010/ )
    • cd19faf : Attempt to isolate a bug by adding a new CHECK
    • 2f48d94 : Set pacer target bitrate to max of bwe and bitrate allocation.
    • a81a239 : Remove DrMemory suppressions for NetEq
    • b136b4f : Roll chromium_revision bce6ee4..b483788 (357989:358023)
    • d6b9d33 : Moves logging of audio effects to when they are enabled
    • 85ec209 : Roll chromium_revision f930f3f..bce6ee4 (357874:357989)
    • 444237e : Remove unused crypto.gni import.
    • cecd7b8 : Disable VP9 resize test.
    • a89c5fb : Roll chromium_revision 1f9589b..f930f3f (357797:357874)
    • 9b66957 : Stop a session when a new connection becomes writable. We cannot do it at the end of sorting because it may stop a session too early. Also remove was_writable_, which is not useful. BUG=webrtc:5119
    • e2a8925 : Move audioproc_test_utils into enable_protobuf condition.
    • 98cc88c : Correctly handle the error case where the CodecId has a negative value
    • 5d4e944 : Revert of Change type of pid_diff (int16_t - uint8_t) according to updates in RTP payload profile. (patchset #3 id:40001 of https://codereview.webrtc.org/1427253002/ )
    • e56c763 : Removing trace checks in VoETestManager.
    • 275d255 : Adding debug dump test.
    • b7a2017 : Roll chromium_revision a8b75a6..1f9589b (357542:357797)
    • 81c5c7f : Change type of pid_diff (int16_t - uint8_t) according to updates in RTP payload profile. Max p_diff is 8 bits.
    • da07290 : Cleanup a few symlinks in setup_links.py
    • f040b23 : Add histograms for send-side delay stats for a sent video stream:
    • c21f0c0 : Remove WEBRTC_ANDROID from hardcoded gtest relative path usage.
    • ff761fb : modules: more interface - include renames
    • 5af9a28 : Roll chromium_revision d131cac..a8b75a6 (357393:357542)
    • 4b938e5 : Hide ACMCodecDB::database_ behind accessors
    • 1fd4a4a : Let AudioCodingModule::SendCodec return MaybeCodecInst
    • 969aeb1 : Revert of Exclude offline bots from CQ config. (patchset #1 id:1 of https://codereview.webrtc.org/1420283013/ )
    • 5ab193f : Remove system_wrappers dep from field_trial_default and metrics_default.
    • de94d08 : Hide ACMCodecDB::codec_settings_ behind an accessor
    • 373284d : Make SendStatisticsProxy outlive ViEChannel.
    • 1ba936a : Revert of Fix for "Android audio playout doesn't support non-call media stream" (patchset #3 id:40001 of https://codereview.webrtc.org/1419693004/ )
    • 0ccae13 : Changed FakeVoiceEngine into a MockVoiceEngine.
    • 5eb9d57 : Re-enable PCAP reading in neteq_rtpplay
    • 32a6eae : Exclude offline bots from CQ config.
    • f1104f6 : Remove TODO referring to issue1981, which I just marked WontFix.
    • 20a3461 : Remove deprecated IsUnresolved() method from SocketAddress API.
    • 22ae293 : Roll chromium_revision 78da654..d131cac (357333:357393)
    • 5a846c0 : Make ConnectionType public in order to add java NetworkObserver.
    • 678c903 : Delete AcmReceiver::SetInitialDelay
    • ce4aef1 : Adding support for simulcast and spatial layers into VideoQualityTest
    • 8cc126f : PRESUBMIT: Enable header guard checks for cpplint.
    • 1d5c9bd : Remove unused method AcmReceiver:RedPayloadType
    • 792982b : Suppress data races in AudioDeviceLinuxPulse::Init.
    • cc41924 : Roll chromium_revision 40d9ba6..78da654 (357298:357333)
    • 6f29a69 : Suppress data races in sctp_close.
    • 0b8d056 : Rename InitCpuFlags suppression.
    • 9bc2667 : ACM/NetEq: Restructure how post-decode VAD is enabled
    • d56d68c : system_wrappers: Fix include header guards.
    • 2b06352 : Roll chromium_revision 0bf3ae4..40d9ba6 (357288:357298)
    • 608213e : Add locks and thread annotations for ReceiverReferenceTimeReportEnabled.
    • a8393d9 : Roll chromium_revision d0662bc..0bf3ae4 (357257:357288)
    • 74f0f35 : Delete a chain of methods in ViE, VoE and ACM
    • e502bbe : Update webrtc/base/common.h after recent _DEBUG-!NDEBUG change.
    • 4040d1e : Roll chromium_revision ca6592b..d0662bc (357129:357257)
    • a41ab93 : Switch usage of _DEBUG macro to NDEBUG.
    • 5c3da4b : Call MediaCodec.stop() on separate thread.
    • 8e1809f : Fix TransientSuppression in audioproc_float
    • 78858d2 : Roll chromium_revision 0ebc3da..ca6592b (357077:357129)
    • 0be3e04 : Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on Android.
    • 8a4f547 : Hang on android when DNS resolution is not done
    • 534dafc : Roll chromium_revision ce45e11..0ebc3da (357029:357077)
    • 102c6a6 : Replace rtc::cricket::Settable with rtc::Maybe
    • bdafe31 : Add aecdump support to audioproc_f.
    • 1367fbd : Roll chromium_revision 657e8d9..ce45e11 (356260:357029)
    • cb3f9bd : Make the nonlinear beamformer steerable
    • 7367463 : Utilize bitrate above codec max to protect video.
    • 315dce7 : Enable VP9 internal resize by default.
    • bbaf363 : Filter overlapping RTP header extensions.
    • 4f5db11 : Make VCMEncodedFrameCallback const.
    • 075fb4b : MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
    • 69ccb33 : Remove redudant encoder rate calls.
    • 4f4f756 : Create isolate files for nonparallel tests.
    • 1295297 : Register header extensions in RtpRtcpObserver to avoid log spam.
    • ee1879c : Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
    • 48ed930 : ACM: Move NACK functionality inside NetEq
    • a35ae7f : Fix chromium-style warnings in webrtc/sound/.
    • 95192fb : Create a 'webrtc_nonparallel_tests' target.
    • 6449990 : Update scalability structure data according to updates in the RTP payload profile.
    • 7464089 : audio_coding: rename interface - include
    • be81fa5 : Rewrote perform_action_on_all_files to be parallell.
    • 32df5ef : Update reference indices according to updates in the RTP payload profile.
    • 1a8240c : Disable P2PTransport...TestFailoverControlledSide on Memcheck
    • b608eb8 : pass clangcl compile options to ignore warnings in gflags.cc
    • e55c42c : Remove limitation on the amount of maximum pending HW decoder inputs.
    • 98f5351 : system_wrappers: rename interface - include
    • ebc0b4e : Use webrtc/base/logging.h for rtp_rtcp.
    • 605db69 : Disable EndToEndTest.AssignsTrans... for memcheck
    • 6408174 : Fix for "Android audio playout doesn't support non-call media stream"
    • 83585c9 : VideoCapturerAndroid: More frequent and verbose logging
    • ec9d187 : Added override keyword to overridden methods to stop compiler warnings.
    • fce4a94 : RentACodec: New class that takes over part of ACMCodecDB's job
    • 77d0d6e : When all connections timed out on writing, delete them all. BUG=5111
    • f116bd0 : Call OnSentPacket for all packets sent in the test framework.
    • f1dcd46 : UBSan: Add blacklist files for WebRTC standalone.
    • 9397d84 : Roll chromium_revision 625f6c8..657e8d9 (356202:356260)
    • 27f6fd3 : Remove noparent from talk/OWNERS.
    • 5ddee02 : Landmine: clobber to remove out/{Debug,Release}/args.gn
    • 4f847da : Use webrtc/base/checks.h in desktop_capture.
    • 85a0496 : Implement AudioSendStream::GetStats().
    • 2a0a2a4 : Add stats for used video codec type for a sent video stream:
    • 18ba3e2 : Roll chromium_revision faa5502..625f6c8 (356073:356202)
    • 18a944b : Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
    • d3b26d9 : Adding the ability to change ICE servers through SetConfiguration.
    • 2b55867 : Exposing DTLS transport state from TransportChannel.
    • b0bb77f : Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1416773003/ )
    • 8f46c63 : Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
    • aed571f : Roll chromium_revision 27af50f..faa5502 (356022:356073)
    • e2a83ee : Introduce rtc::ArrayViewT, which keeps track of an array that it doesn't own
    • ac9d92c : Adding the ability to create an RtpSender without a track.
    • 4cba4eb : Disable denoising for VP9 by default.
    • 65e7d4c : Remove CanCreateAndDestroyManyVideoStreams.
    • c4ef143 : Revert "Add GN Build file for rtc_sound target."
    • 717432f : Remove network_enabled_crit_ in call.cc.
    • 09b38f3 : Re-enable VP9 resize test.
    • 7ef0553 : Fix for Win GN Build.
    • 2d3747d : Fix for Mac GN BUILD.
    • e9eca8f : Removing AudioCoding class, a.k.a the new ACM API
    • f054819 : Add GN Build file for rtc_sound target.
    • 213b598 : Roll chromium_revision c86a4e2..27af50f (356002:356022)
    • 415d2cd : Use webrtc/base/logging.h for video.
    • f9af108 : Roll chromium_revision c708f39..c86a4e2 (355993:356002)
    • 484e548 : Roll chromium_revision bbfaf80..c708f39 (355989:355993)
    • eb2a91e : Roll chromium_revision 5512fa0..bbfaf80 (355985:355989)
    • 7542ed6 : Roll chromium_revision da8662f..5512fa0 (355980:355985)
    • 9ec27e1 : Roll chromium_revision da9833c..da8662f (355969:355980)
    • 5d9b92b : Update Bind to match its comments and always capture by value. Also update the generated count to 9 args.
    • 2dd8bf8 : Roll chromium_revision 53f0e22..da9833c (355953:355969)
    • 7d35afd : Roll chromium_revision bd99556..53f0e22 (355580:355953)
    • 401fb06 : SurfaceTextureHelper: Remove use of quitSafely() because it's API lvl 18
    • 238b15d : SurfaceViewRenderer: Remove use of quitSafely() because it's API lvl 18
    • c3402fc : EGL10.eglCreateWindowSurface(): Replace Surface input with SurfaceHolder
    • 90d67dd : Remove two more deprecated methods from SocketAddress API.
    • 49e196a : Remove VideoFrameType aliases for FrameType.
    • a99069d : Fix win32 header include order in rtp_utility.h.
    • 225789d : Move logging CriticalSection into implementation.
    • aa04299 : Don't wait until distant future to shut down video app.
    • 27dfe20 : Remove final from rtc::Buffer.
    • 1e737c6 : Fix thread safety in VcmCapturer.
    • bbe876f : Set send times in send time history via OnSentPacket.
    • 9a4cd87 : Add support for handling reordered SS data on the receive-side for VP9.
    • a3587fb : clean up field_trial_default target, to be used by remoting_perftests.
    • 00507f8 : Separate StunProber::Start into Prepare and Run so we could create multiple of them and send out STUN pings at regular interval.
    • 4f6a8b5 : Revert of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1406153005/ )
    • e26ce1b : Reland of Add experiment on weak ping delay during call set up time (patchset #1 id:1 of https://codereview.webrtc.org/1423443002/ )
    • 8c425aa : Android: Replace EGL14 with EGL10
    • ff134eb : talk: Use NDEBUG macro.
    • e3422c1 : Remove __DATE__ and __TIME__ from tracing
    • c80741f : Fixing some issues with the direction attribute of m-lines in offers.
    • 56149e5 : Roll chromium_revision 7c002e5..bd99556 (355518:355580)
    • b7edb88 : Prevent BWE rampdowns without new loss reports.
    • 4a859fd : Roll chromium_revision 2c4120b..7c002e5 (355476:355518)
    • 797ef12 : Added StopAecDump function to PeerConnectionFactory.
    • a74c08d : Move i420 files to the right location
    • 48e66b4 : GN: Remove build_overrides/v8.gni
    • 4f4ec0a : Re-Land: Implement AudioReceiveStream::GetStats().
    • b1ce663 : Allow encoders to fall back dynamically to software.
    • b788bc2 : Add Mac-specific resource to modules_unittests.isolate
    • 93ea78b : Add test resources to libjingle_media_unittest.isolate
    • 9589e2a : Update isolate files for swarming tests
    • 4f47ed4 : Roll chromium_revision fecea52..2c4120b (355266:355476)
    • 522fac7 : Roll chromium_revision 9a72f7c..fecea52 (355218:355266)
    • affa39c : Remove time constraint on first retransmit of a packet.
    • c96df77 : - Introduce internal classes WebRtcAudio[Send|Receive]Stream in WebRtcVoiceMediaChannel. - Remove WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer - Create webrtc::AudioSendStreams.
    • f4d23b2 : Remove MockVideoCapturer
    • dfa2815 : Update receive report SSRCs on RemoveSendStream.
    • 0c478b3 : Rename ChannelGroup to CongestionController and move to webrtc/call/.
    • edcbd56 : Adding the OnePlus 2 device to AEC and NS blacklists.
    • 0a87ffc : Fix bug in how send timestamps are converted to 24 bits.
    • e378702 : ChannelGroup cleanup.
    • 45c136b : Adds support for Bluetooth headsets to the iOS audio layer.
    • 6e58720 : Introduce rtc::MaybeT, which either contains a T or not.
    • b64a32b : Remove old VideoFrame::Reset.
    • 3b7c793 : New DtlsIdentityStoreInterface::RequestIdentity added that takes rtc::KeyParams. The old RequestIdentity still exists that take rtc::KeyType.
    • a01d440 : Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
    • 86b0160 : Add stats for average QP per frame for VP8 (for received video streams):
    • 47dcb23 : Roll chromium_revision 01cbe8a..9a72f7c (355025:355218)
    • fcab1cd : Disable VP9 resize test for now.
    • e4f9650 : Remove system_wrappers/interface/trace_event.h
    • 3cf20ed : Will re-enable after libvpx roll, needs to be updated.
    • 0a617e2 : Remove the default send channel in WVoE.
    • 3bf69b1 : Add experiment on weak ping delay during call set up time
    • 30a5b5e : passing |buffer| by reference in AndroidVideoCapturer::OnIncomingFrame
    • 3866c4f : Testing that waiting for a condition variable waits.
    • 43e83d4 : Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
    • 5a197dd : Remove files added by mistake.
    • a457752 : Implement AudioReceiveStream::GetStats().
    • 5460f9b : Workaround for false positive -Wmaybe-uninitialized being triggered on some compilers
    • 9172234 : Roll chromium_revision 5d149df..01cbe8a (354955:355025)
    • 7ae9262 : Suppress libyuv::TestCpuFlag races.
    • eff0fc6 : Adding missing stats class registration, lost in #10298.
    • da535c4 : Add histogram for percentage of sent frames that are limited in resolution due to bandwidth: - "WebRTC.Video.BandwidthLimitedResolutionInPercent"
    • 42b1478 : Roll chromium_revision 0f9b2fd..5d149df (354799:354955)
    • 1897f77 : Make the high frequency correction range depend on the target angle
    • 4a66e4a : Make the separation between target and interferer scenario depend on microphone spacing in NonlinearBeamformer
    • c6aec4b : Fix HW video codec stack traces reporting.
    • 0d97d53 : Fix off-by-one error in PRNG.
    • 86721fe : Roll chromium_revision 8801af5..0f9b2fd (354718:354799)
    • dd2bd26 : Update iOS merge script.
    • 023f3ef : Create network change notifier and pass the event to NetworkManager
    • 0dbf009 : Remove the video channel id completely.
    • f56eca0 : Remove dummyinstantiation.cc.
    • 65a0367 : Revert of Add screenshare perf tests with lossy links (patchset #1 id:1 of https://codereview.webrtc.org/1409513005/ )
    • 5a28939 : Added thread checker to webrtc::Call.
    • 22993e1 : Unify FrameType and VideoFrameType.
    • 4306fc7 : Add histogram for percentage of sent frames that are limited in resolution due to quality: - "WebRTC.Video.QualityLimitedResolutionInPercent"
    • a20de20 : Move ownership of receive ViEChannel to VideoReceiveStream.
    • 1c49bf9 : Roll chromium_revision 37d6b63..8801af5 (354704:354718)
    • 2f24175 : Roll chromium_revision 406df39..37d6b63 (354701:354704)
    • 1760ba3 : Roll chromium_revision d2e1b51..406df39 (354694:354701)
    • 0d2e70e : Roll chromium_revision 54d0dc5..d2e1b51 (354685:354694)
    • df200d1 : Suppressing TestUdpReadyToSendIPv4 on ASan.
    • 743f50a : Roll chromium_revision a3402c9..54d0dc5 (354680:354685)
    • ecce2ba : Roll chromium_revision ae74f84..a3402c9 (354660:354680)
    • 3eab10d : Add back an override of RestoreOriginalPacket.
    • 89f168a : Roll chromium_revision 238710d..ae74f84 (354489:354660)
    • 45daf7b : Implement new version of the NonlinearBeamformer
    • c7a8b08 : Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
    • 9781152 : Add new Android camera events.
    • be16f79 : Remove simulcast bitrate modes.
    • 6ca1ac4 : Get rid of deprecated HttpClient fail_redirect accessors.
    • 861c55e : Transport sequence number should be set also for retransmissions.
    • 1adce14 : Old config events are no longer removed from RtcEventLog.
    • 6a14b9d : Roll chromium_revision 9ce7331..238710d (354427:354489)
    • 12f6802 : Revert "Prepare MediaCodecVideoEncoder for surface textures."
    • 757077b : Removing the TFRC Rate Control
    • 949c2f0 : Move ownership of send ViEChannels and ViEEncoder to VideoSendStream.
    • 112a3d8 : Added functions on libjingle API to start and stop the recording of an RtcEventLog.
    • f85efae : Roll chromium_revision ac4ebe0..9ce7331 (354310:354427)
    • cbc9507 : Temporarily rename P2PTestConductor.
    • a368329 : Roll chromium_revision c40535a..ac4ebe0 (354244:354310)
    • 5e97fb5 : Don't create remote streams if m-line direction doesn't include "send".
    • af1b59c : Cleaning up peerconnection_unittest.
    • 65e15ba : Add native-handle support for single VP8 streams.
    • 4af0f1a : Add screenshare perf tests with lossy links
    • c1aeaf0 : Wire up packet_id / send time callbacks to webrtc via libjingle.
    • 543b6ca : Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/
    • 27576e0 : Landmines support to ease clobbering builds
    • a2f30de : Log Call {audio, video} stream deletions.
    • 5bdddf9 : Move PRNG from BWE test framework to webrtc/test.
    • 411d234 : Roll chromium_revision 6919ce3..c40535a (354197:354244)
    • ab73d13 : Remove internal encoders from VCMCodecDatabase.
    • 6435c1f : Roll chromium_revision c1463d5..6919ce3 (354087:354197)
    • d59daf8 : Merging BaseSession code into WebRtcSession.
    • 8321040 : Roll chromium_revision 760c558..c1463d5 (354059:354087)
    • a9046d0 : Add unit test to decode to a surface texture. Also parameterise on PeerConnectionParameters to prepare for more test variations. (capture and encode to textures)
    • 9f378cd : Roll chromium_revision b8ff103..760c558 (353988:354059)
    • ab9b2d1 : Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
    • 65220a7 : Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
    • bd7de0c : Delete full-band mode from the iSAC codec
    • 1d7bcd8 : Roll chromium_revision 159828f..b8ff103 (353696:353988)
    • 6d387c0 : Android MediaCodecVideoDecoder: Limit max pending frames to number of input buffers
    • 06b869f : Delete iSAC-fb from NetEq
    • 457a61d : Pause/resume pacer from Call instead of via SendStreams.
    • b79472a : Roll chromium_revision c089d37..159828f (353662:353696)
    • fc648b6 : Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
    • 97c3929 : Moving MediaStreamSignaling logic into PeerConnection.
    • a0751c5 : Cleanup OWNERS of talk/app/webrtc.
    • 7dc39f3 : Avoid data race in RtcpReceiver.
    • 73f44f6 : VideoCapturerAndroid, only you SurfaceViewHelper when capturing to textures. SurfaceViewHelper requires EGL14 that was added in API level 17. Since the SurfaceViewHelper is only neeed when we capture to textures, this cl change back to not use it when we are capturing to byte buffers.
    • 9ea2147 : Delete iSAC-fb from AudioCodingModule
    • ec93628 : Fix use of scaler in MediaCodecVideoEncoder
    • 1ac5614 : Remove default receive channel from WVoE; baby step 3. Get rid of default receive channel.
    • 8fb30c3 : Remove default receive channel from WVoE; baby step 2. Rename voe_channel_ to default_send_channel_id_.
    • b8fd39c : Roll chromium_revision 1e0ef42..c089d37 (353553:353662)
    • 3402bcd : Make the WARN_UNUSED_RESULT macro match the Chromium one.
    • 325d414 : Add option to print peer connection factory Java stack traces.
    • a5b62d9 : Replace API v23 calls.
    • e2524ba : Roll chromium_revision 8fec661..1e0ef42 (353502:353553)
    • fc95084 : Fix: RefCountInterface: Make AddRef() and Release() const
    • 52a30e3 : Reland of Android: Put common native VideoFrameBuffer implementation in androidvideocapturer_jni (patchset #1 id:1 of https://codereview.webrtc.org/1389283003/ )
    • 1b40a9a : RefCountInterface: Make AddRef() and Release() const
    • 7a975f7 : Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
    • d940880 : Roll chromium_revision bb79186..8fec661 (353481:353502)
    • 9075417 : Prepare MediaCodecVideoEncoder for surface textures. This make small refactorings to MediaVideoEncoder to prepare for adding support to encode from textures. The C++ layer does not have any functional changes. - Moves ResetEncoder to always work on the codec thread - Adds use of ThreadChecker. - Change Java MediaEncoder.Init to return true or false and introduce method getInputBuffers. - Add simple unit test for Java MediaCodecVideoEncoder.
    • 962c26b : Revert of Android: Put common native VideoFrameBuffer implementation in androidvideocapturer_jni (patchset #1 id:1 of https://codereview.webrtc.org/1391403004/ )
    • 6047221 : Android: Put common native VideoFrameBuffer implementation in native_handle_impl.cc
    • 747c1bc : Android SurfaceTextureHelper: Replace API 21 with API 11 version of setOnFrameAvailableListener()
    • e9e3668 : Android: Add helper function to synchronously execute Callables on Handler
    • 3ba4e28 : Roll chromium_revision 2f90dd6..bb79186 (353472:353481)
    • 450d883 : Roll chromium_revision df7fc05..2f90dd6 (353471:353472)
    • 75926e1 : Roll chromium_revision c3ecc9f..df7fc05 (353465:353471)
    • 6920003 : Roll chromium_revision ce77a26..c3ecc9f (353463:353465)
    • 6fc3f90 : Roll chromium_revision 9c8e128..ce77a26 (353453:353463)
    • 9a717b4 : Roll chromium_revision daae8f2..9c8e128 (353255:353453)
    • 69ddaef : Revert "Add option to print peer connection factory Java stack traces."
    • b68c599 : Add option to print peer connection factory Java stack traces.
    • d4cec0d : Remove MediaChannel::SetRemoteRenderer(). This is following discussion in: https://codereview.webrtc.org/1385893002/diff/60001/talk/media/webrtc/webrtcvoiceengine.cc#newcode2410
    • e5295aa : Roll chromium_revision c4d86a0..daae8f2 (353209:353255)
    • 98c6886 : - Remove AudioTrackRenderer. - Remove AddChannel/RemoveChannel from AudioRenderer interface.
    • 4bac9c5 : Change SetOutputScaling to set a single level, not left/right levels.
    • 0b67546 : Remove default receive channel from WVoE; baby step 1.
    • fb6c02d : Roll chromium_revision 5109f35..c4d86a0 (353030:353209)
    • fd20bb3 : Revert "Allow to print Java stack traces in Android camera, renderer and media codec."
    • 2b298de : Reset media codec thread when Encoder/Decoder object is created.
    • e7f6b56 : VP9: Enable multi-threading for SVC.
    • eefbc3b : Revert of Remove AudioTrackRenderer (patchset #3 id:40001 of https://codereview.webrtc.org/1399553003/ )
    • f0159a7 : Allow to print Java stack traces in Android camera, renderer and media codec.
    • 1c0bb38 : - Remove AudioTrackRenderer. - Remove AddChannel/RemoveChannel from AudioRenderer interface.
    • 69f5760 : Added parsing of either space or colon for sctp-port.
    • e76fb36 : Android SurfaceViewRenderer: Add tests for onMeasure()
    • bf2004b : Android SurfaceViewRenderer: Only clear image in release() if initialized
    • 4e57247 : Provide RSA2048 as per RFC
    • 301aaed : Update to the RtcEventLog protobuf to remove the DebugEvent message.
    • 8ac544e : Get rid of deprecated SocketAddress::IsAny() method.
    • c671139 : Removing M API call for now to green up downstream.
    • ac30642 : Native changes for VideoCapturerAndroid surface texture support These are the necessary changes in C++ related to the video capturer necessary to capture to a surface texture. It does not handle scaling / cropping yet though.
    • 4382d80 : Android SurfaceViewRenderer: Allow to re-init after release() has been called
    • 6ffc330 : Remove references to libpeerconnection.
    • 87f83a9 : Adding support for simulcast and spatial layers into VideoQualityTest
    • 650c994 : Roll chromium_revision 2a2c52e..5109f35 (353013:353030)
    • 3d06eca : Add support to Capture to a texture instead of memory.
    • c1cc854 : Fixing perf regression caused by refactoring full stack tests
    • e23e737 : Disable pacer disabling.
    • 335204c : Revert of Provide RSA2048 as per RFC (patchset #9 id:200001 of https://codereview.webrtc.org/1329493005/ )
    • 0df3eb0 : provide RSA2048 as per RFC
    • f839dcc : Add stats for rendered pixels (sqrt(w*h)) per second: - "WebRTC.Video.RenderSqrtPixelsPerSecond"
    • e0e88cf : Roll chromium_revision 22ce537..2a2c52e (352889:353013)
    • e78e2c7 : Using different sequence numbers for different SSRCs
    • fddf6e5 : Use WebRTC logging in MediaCodec JNI code.
    • 21622a1 : Add option to print peer connection factory Java stack traces.
    • 91b348c : Android MediaCodecVideoDecoder: Manage lifetime of texture frames
    • 87962a9 : Roll chromium_revision df4218d..22ce537 (352811:352889)
    • c7199c2 : Read the number of TLs for VP9 too + cleanup
    • 7854328 : Fix minor GYP error in webrtc/tools/internal_tools.gyp
    • 172f009 : Get rid of SCHANNEL code.
    • 0a6380f : Roll chromium_revision d47c242..df4218d (352743:352811)
    • 70a5e0e : Remove (u)int typedefs from basictypes.h.
    • 0c4e06b : Use suffixed {uint,int}{8,16,32,64}_t types.
    • 8d15bd6 : Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
    • d97ec30 : Remove default receive channel from WVoE; baby step 0.
    • 67bcb60 : GN: Port frame_analyzer and rgba_to_i420_converter targets
    • 7f315f5 : Roll chromium_revision 6ebca7d..d47c242 (352640:352743)
    • a38e31a : Update lower-level codereview.settings files.
    • a10492f : Fix VS 2015 warning by adding an additional cast
    • 4139c0f : Java binding for RtpSender/RtpReceiver.
    • 1095069 : Revert "Transport sequence number should be set also for retransmissions."
    • a295320 : Roll chromium_revision 2a1b03e..6ebca7d (352576:352640)
    • 0a6c4ca : Catching more errors when parsing ICE server URLs.
    • 8104479 : Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ )
    • 092b133 : Collecting encode_time_ms for each frame.
    • af4ced9 : Transport sequence number should be set also for retransmissions.
    • 86fa298 : Roll chromium_revision 4bf3678..2a1b03e (352512:352576)
    • 5d0379d : Remove kSkipFrame usage.
    • 13c433c : Add delay metric (includes network delay (rtt/2) + jitter delay + decode time + render delay): - "WebRTC.Video.OnewayDelayInMs"
    • 7bd242e : Enabling screensharing tests for Android
    • 9359b5b : Disabling AudioDeviceTest.StartStopPlayout on Android.
    • d89f82a : Roll chromium_revision c511263..4bf3678 (352322:352512)
    • 09f1350 : Add option to reset Android video renderer first frame flag.
    • 6caafbe : Convert uint16_t to int for WebRTC cipher/crypto suite.
    • 1b33da1 : SurfaceTextureHelper fixes
    • 4185032 : Add ThreadChecker class to ThreadUtils
    • d2838a7 : Roll chromium_revision 07b4a8e..c511263 (352281:352322)
    • e0bce24 : VideoCapturerAndroid: Add custom nativeCreateVideoCapturer()
    • 723dff1 : Poll stats more often to get more stable stats in ramp-up tests.
    • 4cd053f : Only catch UnsatisfiedLinkError in Logging.java.
    • f3a7c9d : In rampup tests, set start time when starting poller thread.
    • 95cd8ea : Enable HW NS for N6 to fix HW AEC issue
    • dec5ebf : Move sent key frame stats to send_statistics_proxy class.
    • 42b6c63 : autoroller: Allow to specify Rietveld e-mail.
    • 990d57d : Fix file order in base.gyp.
    • ba0f0a5 : Disable flaky WebRtcVideoChannel2BaseTest.* on DrMemory/memcheck.
    • 4bd8d09 : Roll chromium_revision ca4c339..07b4a8e (352257:352281)
    • 96a70f0 : Exclude WebRtcVideoChannel2BaseTest.AddRemoveCapturerMultipleSources on Dr Memory.
    • b5fd46e : Exclude WebRtcVideoChannel2BaseTest.AddRemoveCapturerMultipleSources on Memcheck
    • 42b4faa : Fix a build issue when use external OpenSSL.
    • 6df1ef6 : Roll chromium_revision 4ce3c08..ca4c339 (352000:352257)
    • bc0938e : Android VideoRendererGui: Make deep copy of incoming texture frames
    • 44bf6f5 : Android MediaCodecVideoDecoder: Split DecoderOutputBufferInfo into DecodedByteBuffer and DecodedTextureBuffer
    • 13b96ba : Adding APM configuration in AEC dump.
    • 371dc7e : WebRtc Win Desktop capture: ignore Win8+ Modern Apps' windows.
    • 913e645 : Loopback and audio only mode.
    • f9c23ca : Exclude WebRtcVideoChannel2BaseTest.GetStats on linux memcheck
    • 9dff0ba : Fix MSVS project files generation.
    • 067fb65 : Roll chromium_revision 7fddcec..4ce3c08 (351973:352000)
    • a050e98 : Avoid race in RampUpTest
    • 7e31937 : Android MediaCodecVideoDecoder: Cleanup to prepare for texture liftime management
    • 6781ea4 : jni/native_handle_impl.h: Move implementation into .cc file
    • 417fec2 : autoroller: Add CQ_EXTRA_TRYBOTS, CQ feature and --skip-cq flag.
    • 401025d : Roll chromium_revision 354cc7d..7fddcec (351828:351973)
    • 1d8a506 : Add a PacketOptions struct to webrtc::Transport.
    • da903ea : Unify newapi::RtcpMode and RTCPMethod.
    • c8ba105 : Roll chromium_revision 681f0cd..354cc7d (351698:351828)
    • a9c584d : autoroller: Always roll and improve description
    • 6c2ba7d : autoroller: Add TBR= field and always update the checkout
    • 18b042f : autoroller: Use HEAD instead of LKGR.
    • 5aaa9b4 : Removed unused API functions in AudioProcessing and AudioProcessingModule
    • 5629a1d : Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004.
    • cf18b34 : Align new VoE API with design.
    • 8c471e7 : Objective-C++ style guide changes for iOS ADM
    • fb30c1b : Update VP8 settings to avoid spending bitrate on static areas. PERF NOTE
    • 5b14b42 : Remove unused SignalMediaError and infrastructure.
    • 49f9cdb : Fix bug where rtcp::TransportFeedback may generate incorrect messages.
    • b09b660 : Remove cricket::VideoFrame::Set/GetElapsedTime()
    • dfc8f4f : Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'.
    • edba998 : Roll chromium_revision 8cf53d6..681f0cd (351112:351698)
    • 0ecf1b2 : Android focus problem on rear camera.
    • 98ab3a4 : Don't link with audio codecs that we don't use
    • 456696a : Reland Change WebRTC SslCipher to be exposed as number only
    • 27dc29b : Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ )
    • 4fe3c9a : Change WebRTC SslCipher to be exposed as number only.
    • 2f8a4ca : Add OWNERS for ObjC dirs.
    • d0b3143 : Do not time out a port if its role switched from controlled to controlling. Also fix some comments. BUG=webrtc:5026
    • 898d21c : WebRTC might leak srflx ip address when multiple_routes disabled and IceTransportType is relay.
    • c4d3a5d : Thinning out the Transport class.
    • 2b342bf : Delete a connection only if it has timed out on writing and not receiving for 10 seconds.
    • 27551c9 : Android RendererCommon: Refactor getSamplingMatrix()
    • 4a8e9c5 : Remove overrides folder.
    • bbda54e : Android MediaDecoder: Use frame pool to avoid allocations for non-surface decoding
    • ee2bf41 : Update build files to use webrtc_overrides in Chromium instead of overrides.
    • 6ba8e4a : ACM: Remove a few local enums that were no longer used
    • d094c04 : Remove AgcManager.
    • a67696b : Reland of Adding PeerConnectionInterface::SetConfiguration method. (patchset #1 id:1 of https://codereview.webrtc.org/1361263002/ )
    • 38778b0 : Add unit test for nack bandwidth constraint.
    • 98db68f : If gather_continually is set to true, keep the last port allocator session running while stopping all existing process of getting ports (when p2ptransportchannel first becomes writable).
    • 24b52f8 : Android GlRectDrawer: Add test for OES texture rendering
    • 1d640e5 : JavaVideoRendererWrapper: Use jlongFromPointer() to convert frame pointer to jlong
    • 63b3454 : Simplify handling of options in WebRtcVoiceMediaEngine. Also removes unnecessary typedef ChannelList.
    • 86fd9ed : Set RtcpSender transport at construction.
    • 38502a7 : Remove isolate_deps_dir from .gitignore
    • 092508a : Fix bug in ramp-up tests stats where rtx was accounted for in the media ssrc.
    • b5815c8 : Revert of Android VideoCapturer: Send ByteBuffer instead of byte[] (patchset #1 id:1 of https://codereview.webrtc.org/1372813002/ )
    • fb9e763 : Remove last use of ACMAMRPackingFormat
    • d6024e3 : Roll chromium_revision 310ea93..8cf53d6 (349094:351112)
    • 70ab1a1 : Exposing RtpSenders and RtpReceivers from PeerConnection.
    • 8e9cb09 : Android: Add unittests for SurfaceTextureHelper
    • 4fa648b : Adding 20-second timeout to Java and Objective-C tests.
    • 8108764 : Analyze support in gyp_webrtc
    • 2d56668 : Unify Transport and newapi::Transport interfaces.
    • 8387c5f : Remove AMR format parameter from AudioCoder in utility
    • 1968d3f : Simplify VCMTimestampMap.
    • 8c404fa : When doing DisableEquivalentPhases, exclude those AllocationSequences whose network has ever been removed. It is unlikely the sockets/ports/candidates created from those AllocationSequences will still be valid.
    • 1f429e3 : Passing the new policy from PeerConnection RTCConfiguration to p2ptransportchannel. This CL does not use the new policy yet. BUG=
    • cb3649b : Android VideoCapturer: Send ByteBuffer instead of byte[]
    • 4b808ee : ACM: Remove unused and deprecated types
    • 1bd0e03 : ACM: Removing runtime APIs related to playout mode
    • d417523 : Minor fix for debug logging on Android
    • 4fbd145 : Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side.
    • d2413e5 : Fix the C++ SurfaceTextureHolder This cl moves back loading java SurfaceTextureHolder to the ClassReferenceHolder and use FindClass through ClassReferenceHolder. Without this, jni-FindClass returns nullptr in surfacetexturehelper_jni.cc.
    • 1ab271c : Android SurfaceTextureHelper: Don't wait for pending frames in disconnect()
    • 3e9eb4b : Add C++ SurfaceTextureHandler This cl adds a C++ counterpart of the Java SurfaceTextureHandler. It can be used for creating a webrtc::VideoFrames from a native handle and also guarantee that the Java SurfaceTexture is notified when the VideoFrame is no longer in use.
    • 82d6f2a : ACM: Remove ACMVQMonCallback object
    • 69984f0 : Fixes logging levels in WebRtcAudioXXX.java classes
    • d6d27e7 : Update isolate.gypi to support Swarming + move .isolate files
    • c97be6a : Disable TestUdpReadyToSendIPv4 under MSan.
    • 4d47aa3 : Fallback to system log when webrtc tracing not enabled.
    • 1741770 : Implement a high-QP threshold for Android H.264.
    • a323fd6 : Removes Nexus 6 from OpenSL ES blacklist.
    • 88799d9 : RTCEAGLVideoView: Fix GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT error.
    • 94c0105 : Added peah@ to the watch lists
    • 702f397 : GN: Do not use forward_dependent_configs_from variable.
    • 5c389d3 : Split webrtc/video into webrtc/{audio,call,video}.
    • 82e2055 : Modifies invalid DCHECK in AudioRecordJni::OnCacheDirectBufferAddress()
    • 44d5d7b : Autoroll: Update checkout before reading chromium_revision.
    • 495d2fd : Reland of "Android GlRectDrawer: Add test for RGB rendering"
    • 3fd7be4 : Revert of Don't link with audio codecs that we don't use (patchset #4 id:60001 of https://codereview.webrtc.org/1349393003/ )
    • a53e383 : Revert of CodecOwner: Don't look at definitions for classes we don't link with (patchset #1 id:1 of https://codereview.webrtc.org/1364233002/ )
    • 67e0cf1 : Android AppRTCDemo: Add slider for changing camera capture quality during call
    • 574d5da : CodecOwner::SetEncoders: Return error code when given bad arguments
    • 6979b02 : Adding stub files for RtpSender/RtpReceiver.
    • 4ba059d : Remove custom handler since the logger already logs to console by default. BUG= R=glaznev@webrtc.org
    • 8937437 : Do not prune if the current best connection is weak. Otherwise, we may delete a useful connection because the current best connection may be failing.
    • ea70d77 : VideoCapturerAndroid: Add test for making calls on stopped camera
    • 59e72ab : Enable logging for Mac by default on debug builds.
    • f4d38ea : CodecOwner: Don't look at definitions for classes we don't link with
    • 34fbfff : Remove VideoMediaChannel::SetRender().
    • 5e9a1bc : Revert of Android GlRectDrawer: Add test for RGB rendering (patchset #3 id:40001 of https://codereview.webrtc.org/1367923002/ )
    • a58ea78 : 1. Add receiving state as part of the connection sorting criteria. So if a connection's receiving state changes, it will re-select a better connection if there is any. This will paves the way for continuous nomination lite and multi-networking. 2. Combined checking and pinging to remove some redundant checking and to make it switch to more frequent ping mode earlier.
    • 8a88dd2 : Stability improvement for audio recording on Android
    • 2bc68c7 : Wire up QualityScaler for H.264 on Android.
    • 7076729 : Enable SurfaceViewRenderer for AppRTCDemo
    • 9236bb1 : Minor fix for improving logging of supported platform effects
    • 6b8d355 : Reland "Wire up send-side bandwidth estimation." Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/
    • 8c266e6 : H264 bitstream parser.
    • 6b20ad9 : Android GlRectDrawer: Add test for RGB rendering
    • 2efe58b : VideoCapturerAndroidTest: Dispose PeerConnectionFactory with pending frames
    • ec249d4 : ACMCodecDB: Remove unused stuff, and move private stuff to anonymous namespace
    • 4a3ccad : Remove SetAudioDelayOffset() and friends.
    • f66a925 : Don't link with audio codecs that we don't use
    • 61e933e : Remove ChannelManager::GetCapabilities()
    • c675ddd : video_capture: Better support for UYVY
    • 74d85e1 : Reduce locking in overuse frame detector now that (as of r9508) the observer_ and options_ can only be set at construction time. E.g. no lock is any longer held while doing the callback.
    • facbbec : Remove use of DeviceManager from ChannelManager.
    • 7603c76 : Revert of Adding PeerConnectionInterface::SetConfiguration method. (patchset #4 id:60001 of https://codereview.webrtc.org/1317353005/ )
    • 70702af : Adding PeerConnectionInterface::SetConfiguration method.
    • 53eee43 : Address the comment from 1367553002.
    • 2e4b620 : TcpPort doesn't connect when calling gmail with non-proxied UDP disabled.
    • cdfe20b : Fix the maximum native sample rate in AudioProcessing
    • cbecd35 : Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
    • d0b5b09 : Add myself as OWNER of webrtc/voice_engine and talk/media/webrtc.
    • 7cf0445 : Remove ViEChannel::StartSend deadlock suppression.
    • b7af7b0 : Add myself to watchlist for a few subtrees of the repo.
    • 8bffba7 : Fix BWE bug where audio has timestamps in us.
    • 6d92bf5 : Returning correct duration estimate on Opus DTX packets.
    • c14f5ff : Improving support for Android Audio Effects in WebRTC. Now also supports AGC and NS effects and adds the possibility to override default settings.
    • c9bbeb0 : Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
    • d5c75b1 : Reduce LS_INFO spam from voice_engine/.
    • 7d17336 : Remove the [Un]RegisterVoiceProcessor() API.
    • 0967734 : Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used.
    • f706c8a : VideoCapturerAndroid: Fix threading issues
    • a81a42f : Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
    • 2d4e6c5 : Fixing camera capture for video_loopback
    • 47ee2f3 : TransportController refactoring.
    • 8967183 : Simple cleanups of AudioDecoder and AudioEncoder classes
    • c1a1b35 : Remove the SetLocalMonitor() API.
    • 07d0936 : Purge nss files and dependencies.
    • 7404368 : Move AudioDecoderIsac* to its own files
    • 7083e11 : Remove callback_cs_ in ViEEncoder.
    • 8212265 : Android: Add class ThreadUtils with helper function joinUninterruptibly()
    • 6faf5be : Move AudioDecoderPcm* next to AudioEncoderPcm*
    • d4818e7 : Using static frame generator when no scrolling
    • 9b5476d : sslidentity.cc/IntKeyTypeFamilyToKeyType function added, converting from int to KeyType.
    • ef165ee : Wire up send-side bandwidth estimation.
    • 22011c1 : Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle).
    • 7317248 : Rename CaptureThread to EncodingThread.
    • ef5d5e4 : Add field trial for automic resize in MediaCodecVideoEncoder.
    • 1356ba5 : Fixing target_bitrate_bps for a FullStackTest
    • 8f4f00f : CQ: Update trybots
    • e4ba6ce : Log the tag in native log stream.
    • ebbf8a8 : Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it.
    • 04ac81f : Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet). BUG=4937 R=pthatcher@webrtc.org
    • 5bfc6cb : Revert "Android: Enable C99 mode instead of C89 (default)."
    • 275a2f1 : Revert of Replace readable with receiving where receiving means receiving anything (stun ping, response or da… (patchset #7 id:340001 of https://codereview.webrtc.org/1351673003/ )
    • ae16f85 : Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet). If a connection does not receive for 30 seconds, it will be deleted. BUG=
    • c19922c : Android SurfaceViewRenderer: Block in release() until frames are returned and cleanup is done
    • e6d3ada : Re-add SurfaceTexture as member for setLocalPreview in VideoCapturerAndroid. The Android camera api requires a surface to be set in order work. In https://codereview.webrtc.org/1354683004/ this surfaceTexture was removed as a member but it turns out that can lead to camera freezes when the device is rotated. This cl re-adds the surface as a member.
    • 40bf493 : Revert of Update build files to use webrtc_overrides in Chromium instead of overrides. (patchset #2 id:20001 of https://codereview.webrtc.org/1354933002/ )
    • 780be75 : Make PeerConnectionTest.doTest wait for ice candidates This change the PeerConnectionTest.doTest wait for at least one ice candidate and also make sure the list of candidates in gotIceCandidates is synchronized.
    • baae0a8 : Update build files to use webrtc_overrides in Chromium instead of overrides.
    • 35d1767 : Remove the video capture module on Android. Video capture for android is now implemented in talk/app/webrtc/androidvideocapturer.h
    • 8902433 : Revert "TransportController refactoring."
    • 9af63f4 : TransportController refactoring.
    • 2803a40 : Fix ChromeOS build (C99 break)
    • 4a78308 : Android: Add helper class GlTextureFrameBuffer
    • e1aa5b5 : This relands "Tool to convert RtcEventLog files to RtpDump format.", commit 35624c2c3686a2ad40daffe073aa78507b0ef88e.
    • ca14b2f : Add system log fallback when native logging is unavailable.
    • e510d7f : Remove ACM AudioCodingFeedback callback object and derived classes
    • be49595 : Revert of Tool to convert RtcEventLog files to RtpDump format. (patchset #11 id:200001 of https://codereview.webrtc.org/1297653002/ )
    • f4aa4c2 : Remove id from VideoProcessingModule.
    • 3520f9e : Removes camera.setPreviewTexture in doStopCaptureOnCameraThread and removes the try catch statement since the only method throwing an exception was setPreviewTexture.
    • 586b19b : Enable probing with repeated payload packets by default.
    • 71df77b : Remove overridden basictypes.h.
    • 061b79a : ACM: Remove functions related to DTMF
    • 11d583f : Fix a bug in RtpFileSource related to RTCP packets in rtpdump files
    • 35624c2 : Tool to convert RtcEventLog files to RtpDump format.
    • 7cbd188 : Remove GICE (again).
    • ac547a6 : Remove channel ids from various interfaces.
    • 1d5198d : Fix parameter in VP9 resize test.
    • f350720 : VP9: Add automaticeResize to codec setting.
    • e1c5ec7 : Fixing bad merge (CHECK is now RTC_CHECK)
    • fdd1b9a : Reland: Bailing out if pc factory fails to get created.
    • b071a19 : Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters.
    • ae856f2 : Added support for logging the SSRC corresponding to AudioPlayout events.
    • 48c46db : Reduces default sample rate from 44.1kHz to 16kHz to ensure that we can open up audio in communication mode also on older devices that only supports it in combination with 16kHz.
    • d2320ce : CQ: Remove baremetal machines from CQ bots.
    • 5d6a06c : Refactoring full stack and loopback tests
    • f2bfc2b : Remove some dead code.
    • e64fbce : Changed loopback transport in RtxNackTest to not store sequence numbers for retransmitted packets.
    • ada4c13 : Move AudioDecoderG722 next to AudioEncoderG722
    • 97395b6 : Remove dependency on Chromium's base/logging.h in diagnostic_logging.h.
    • 91d6ede : Add RTC_ prefix to (D)CHECKs and related macros.
    • c0ac6ca : Move AudioDecoderPcm16B next to AudioEncoderPcm16B
    • b697cea : Roll chromium_revision 5482f56..310ea93 (347609:349094)
    • fff9f17 : Move AudioDecoderIlbc next to AudioEncoderIlbc
    • 1f9baab : Remove the preprocessor symbol WEBRTC_CODEC_AVT (it was always defined)
    • 7754285 : Log to the webrtc log stream from webrtc/modules java code. The purpose is to gather all webrtc logging in a single place and allow the app to redirect all webrtc logging to a single stream for offline debugging.
    • 2520e72 : VP9: Enable static threshold for non-screen content.
    • 5975b3c : Log to webrtc logging stream from java code. Future log messages should all be sent to org.webrtc.Logging as well.
    • eecbab7 : Roll chromium_revision a28d8d5..5482f56 (346100:347609)
    • 844a910 : Remove the preprocessor symbol WEBRTC_CODEC_PCM16 (it was always defined)
    • 3841943 : Consolidate constructormagic macros with Chromium version and remove Chromium override.
    • 3c089d7 : Add RTC_ prefix to contructormagic macros.
    • afb6b5e : Ensure all test targets are built on Android.
    • 8dba03d : Temporarily define RTC_DISALLOW_ASSIGN in Chromium constructormagic override.
    • 207370f : Android MediaCodecVideoDecoder: Remove redundant useSurface arguments
    • 01ddf01 : Revert of Bailing out if pc factory fails to get created. (patchset #1 id:1 of https://codereview.webrtc.org/1339923004/ )
    • 6eb75d9 : Bailing out if pc factory fails to get created.
    • 2338fec : Partial revert of r9936.
    • 32b5d23 : Add an option to avoid Java video track release when peer connection is closed.
    • ebed24d : Do not print C++ line numbers for Java logging.
    • 0b05879 : Move AudioDecoderOpus next to AudioEncoderOpus
    • ec0feb6 : Add --skip-try flag to autoroll script.
    • 8e4e8b0 : Simplify BitrateAllocator::AddBitrateObserver.
    • dcb8998 : Keep lock after updating encoder parameters.
    • a753177 : Remove default ViEEncoder encoder instance.
    • 709ed67 : Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
    • 4ae28a1 : Android: Add SurfaceTextureHelper for creating and managing SurfaceTextures
    • 7bff85c : Android: Enable C99 mode instead of C89 (default).
    • 66f0da2 : Log to webrtc logging stream from java code. Future log messages should all be sent to org.webrtc.Logging as well.
    • 73a93e8 : Add a ParseHeader method to RtcpPacket, for parsing common RTCP header.
    • 1cb121d : Reset frame timestamp epoch for new capturers.
    • 6304626 : Add a rate tracker that tracks rate over a given interval split up into buckets that accumulate unit counts for their portion of said interval and use this instead of the standard rate tracker so that the values of retrieved frame rate stats are completely independent of the polling rate.
    • 2b18084 : Only allow static strings as ProcessThread names.
    • ea06a58 : Android video capture: Remove duplicated code and fix spelling mistakes
    • fc9dd17 : Added boundary check for array access as a short-term way of fixing the bug of out-of-bounds reads into the array
    • 5e023eb : Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor
    • 36d619b : Log timestamps when old frames are delivered.
    • 847855b : Add a name to the ProcessThread constructor.
    • c32d2db : Refactor RTPPacketHistory to use a packet struct.
    • 92068ee : Android: Guard against switching camera on stopped camera
    • c68700f : Add README.md to root directory
    • f08fb29 : Disable flaky test (WebRtcVideoChannel2Base.AddRemoveCapturer) on Dr. Memory.
    • c06a716 : Android: Add new renderer SurfaceViewRenderer
    • df1a171 : Remove unused event in video_capture_input.cc.
    • d12140a : Revert change which removes GICE.
    • fab882b : Remove obsolete typingmonitor.cc/.h files.
    • 4ed3658 : Avoids crashes in Java-based InitRecording().
    • 39720f2 : ACM CodecOwner: Test that we reset speech encoder when enabling CNG or RED
    • 9b66d3b : MockAudioEncoder: Use a dedicated marker method for test expectations
    • 1dd98f3 : - Rename VoiceChannel::MuteStream() - SetAudioSend() (incl. media channel) - Rename VideoChannel::MuteStream() - SetVideoSend() (incl. media channel) - Collapse NnChannel::SetChannelOptions() into the above. - Collapse VoiceChannel::SetLocalRenderer into SetAudioSend().
    • 9a78d22 : Revert of Consolidate constructormagic macros with Chromium version and remove Chromium override. (patchset #4 id:60001 of https://codereview.webrtc.org/1316363005/ )
    • 0de8ff4 : Consolidate constructormagic macros with Chromium version and remove Chromium override.
    • 11e4985 : GN: Fix iOS build.
    • c2db810 : Remove VideoRendererInterface::CanApplyRotation()
    • f6901b0 : Remove NullVideoFrame
    • 8ce0bd5 : Android video rendering: Fix texture matrix multiplication order
    • 942a699 : AudioEncoderOpusTest.PacketLossRateOptimized: Fix bug and make prettier
    • 2feafdb : Enable automatic resizing for RTX-enabled senders.
    • 529528c : Android video rendering: Apply SurfaceTexture.getTransformationMatrix()
    • 66f4339 : Remove [Voice|Video]MediaChannel::GetOptions().
    • c99ebc1 : Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize
    • d944067 : Disable flaky test (WebRtcVideoChannel2Base.GetStatsMultipleSendStreams) on Dr. Memory.
    • b04965c : Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.
    • 3f5f1c2 : Change return type of AudioEncoder::SetMaxPlaybackRate to void
    • e9e7896 : Turn webrtc::Vad into a pure virtual interface
    • 233bd87 : Add RemoteEstimatorProxy for capturing receive times
    • dd2ca84 : Fix name for NDK protobuf libraries.
    • 66c42df : Alphabetize common_audio/OWNERS.
    • 7764973 : Add magjed@ as owner for talk/app/webrtc/androidtests/ and talk/app/webrtc/java/jni/
    • 76b3147 : Disable flaky WebRtcVideoChannel2Base, EndToEndTest tests on Dr. Memory.
    • 12cfc9b : Fold AudioEncoderMutable into AudioEncoder
    • cd3c475 : Updating common_audio/OWNERS
    • 68786d2 : Wire up PacketTime to ReceiveStreams.
    • e526974 : Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/
    • a9839dd : Use of override keyword to fix chromium trybot
    • 04ada47 : Add third_party/lss and third_party/proguard to .gitignore.
    • f325d21 : Disable VideoSendStreamTest.VP9FlexMode.
    • c3aa12d : Add utility class for unwrapping 16 bit sequence numbers
    • caa5f4b : Update to the neteq_rtpplay utility to support RtcEventLog input files.
    • f3ecdb9 : Replacing SSLIdentity* with scoped_refptrRTCCertificate in TransportChannel layer.
    • 8006f07 : Remove unused TypingMonitor class.
    • 7f6a6fc : Enabling spatial layers in VP9Impl. Filter layers in the loopback test.
    • e313e02 : Remove unnecessary fields from VoE SharedData.
    • 746210f : Remove unused overuse detection metric (capture jitter).
    • 3dfe5d3 : Remove arraysize.h gcc hack and Chromium override.
    • e9ad18b : Remove obsolete soundclip.cc/.h files.
    • 1c7d48d : Let max default bitrate depend on resolution when configuring one video stream (was previously always 2Mbps).
    • 6322467 : PRESUBMIT: Exclude some files from 80-character limit check.
    • 81db11a : copy-red: Fill an rtc::Buffer with bytes the easy way
    • 86d907c : Refactor the AudioDevice for iOS and improve the performance and stability
    • 05cfcd3 : Full stack graphs
    • 110443c : Fix for frame resolution in encoded frame callback.
    • 7b38f69 : Add placeholder files for talk/app/webrtc/mediacontroller.cc/.h to be able to update Chrome's libjingle.gyp before the MediaController implementation CL is submitted.
    • c0c7d2e : GN: Fix invalid configuration for Android GCC build.
    • bb741b3 : Remove GetOutputScaling from VoiceMediaChannel.
    • 0ab8ca8 : Remove x11 from libjingle_media
    • 88703d7 : Disable base/logging.h stderr logs by default for webrtc/ tests.
    • 9eb1365 : Revert of purge nss files and dependencies (patchset #1 id:1 of https://codereview.webrtc.org/1313233005/ )
    • fd4df46 : Fix build when using Xcode 7 which contains .tbd files instead of .dylib
    • d5ae6ae : Fix ScreenCapturerWinGdi to handle DesktopFrameWin::Create() errors.
    • 3cc834a : Add more IceCandidatePairType for host-host CandidatePair
    • 250bdc7 : Exclude VideoSendStreamTest.VP9FlexMode on linux_memcheck.
    • 5647a2c : purge nss files and dependencies
    • e7a0de7 : CameraEnumerationAndroid: Add getSupportedFormats() implementation using android.hardware.camera2
    • 242d638 : VP9 codec controls for screensharing
    • 318673c : Update SendTimeHistory to store complete PacketInfo, not just send time
    • c8a1ccc : Fixed base time in TransportFeedback message writing.
    • d415629 : Remove AsyncHttpRequest, AutoPortAllocator, ConnectivityChecker, and HttpPortAllocator.
    • 2f9fd5d : Changed LogRtpHeader to read the header length from the packet instead of requiring an extra argument.
    • b6b0b92 : Rate limit the low bandwidth / min bitrate warning to once every 10 seconds.
    • be9b7b6 : Make sure ByteReader and ByteWriter classes (and their specializations) don't perform operations that have implementation-specific or undefined behavior.
    • 47d78cc : Pass the encoder's internal source property through to video_sender to request a keyframe from the external encoder
    • dfbe679 : Cleanup: Remove duplicated functions
    • 9743d07 : Reland "Adding unittests to AudioConferenceMixer."
    • 6ee69aa : Add scrolling screenshare test to full_stack perf tests.
    • 658910c : Revert "Speculative revert of "- Move test cases for more natural ordering.""
    • 7fabd46 : Don't set V bit in flexible mode
    • 7afc12f : VideoRendererGui: Move to async rendering and remove no longer needed code
    • 4df08ff : GN: Fix compilation with NaCl toolchain
    • 1a591dd : Android GlUtil: Add helper functions generateTexture/deleteTexture
    • 6aae757 : On FATAL, log which unsupported encoder the caller wanted us to create
    • ed4224f : Android GlRectDrawer: Add fragment shader for RGB(A) textures
    • d4563f4 : Revert of Excluding two troublesome trybots from the CQ config. (patchset #1 id:1 of https://codereview.webrtc.org/1310953006/ )
    • 71cfe69 : For TestResolverShutdown, use address that can't be resolved.
    • e63d2a1 : Add JNI/java wrapper for the file rotating logging class.
    • abd0d1a : Handle all RTCICEConnectionState values in ARDVideoCallViewController
    • 4d2f4d1 : - Make shared EGL context used for HW video decoding member of decoder factory class. - Add new Peer connection factory method to initialize shared EGL context.
    • c36d4df : Use committer list from chrome-infra-auth
    • 098c1de : Fixes for PNaCl build of remoting client plugin with GN.
    • 97579a4 : Add option to enable ECDSA key for Java API.
    • eebc099 : Add magjed@ as owner for talk/app/webrtc/java/android/org/webrtc/
    • 194ccea : Do not use HW H.264 encoder on Nexus 7.
    • 4edc39c : Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter.
    • 0f9af01 : Added send stream test case for VP9 header.
    • fa7cb8e : Excluding two troublesome trybots from the CQ config.
    • 02d283a : Speculative revert of "- Move test cases for more natural ordering."
    • 05f71fc : NetEq: Fixing a corner case with depleted sync buffer
    • 521875a : Use RtcpPacket to send APP in RtcpSender
    • e551f12 : Revert "Adding unittests to AudioConferenceMixer."
    • 22c2729 : Adding unittests to AudioConferenceMixer.
    • b7306ae : Revert "Avoiding size_t in MIPS."
    • 32e2f46 : Avoiding size_t in MIPS.
    • 2c27430 : Print some output in long perf tests, to keep them alive
    • 0f4b373 : Stylizing AudioConferenceMixer.
    • ca28fdc : Use RtcpPacket to send XR (RTRR, DLRR, VOIP) in RtcpSender
    • c252dab : CameraEnumerationAndroid: Make getSupportedFormats() an interface
    • c92c23d : Roll chromium_revision f8d6ba9..a28d8d5 (337800:346100)
    • c20a5dc : - Move test cases for more natural ordering. - Get rid of the CoInitialize tests for WVoE/WViE.
    • 3c4ef29 : NetEq: Allow negative shift in BackgroundNoise::SaveParameters
    • 3a14bf3 : Replacing SSLIdentity* with scoped_refptrRTCCertificate in the cricket::TransportDescriptionFactory layers.
    • a6cba3a : Java VideoRenderer.Callbacks: Make renderFrame() interface asynchronous
    • 1380e26 : Convert some more things to size_t.
    • e8386d2 : Added send-thresholding and fixed text packet dumping. Also a little squelch for the over-max-MTU log spam we see in there.
    • 79de90b : Do not explicitly delete OpenGL shaders in VideoRendererGui.
    • f42376c : Wire up currently-received video codec to stats.
    • 6813ec8 : VideoCapturerAndroid: Move to android folder and split out camera enumeration into separate file
    • 9e69abf : Added logging using the raw variant of the new aec logging macros
    • 4fbae2b : Add send transports to individual webrtc::Call streams.
    • 6480d03 : Make jni_helpers build on arm32.
    • 6ec1f92 : AndroidVideoCapturer: Delegate framerate choice to VideoCapturerAndroid.java
    • 98f3cc5 : NetEq: Removing two asserts
    • 1e346b2 : NetEq: Minor follow-up fix in StatisticsCalculator
    • 116c84e : NetEq: Fixing a bug that caused rtc::checked_cast to trigger
    • 9c3efd0 : Reland: Implement NetEq's CurrentDelay function
    • a567bf3 : Rename local variable to avoid shadowing
    • 3154568 : Using 'override' keyword in dtlstransport.h. Chromium FYI trybots were complaining about virtual methods not being marked override.
    • 4376648 : AudioDecoder: Replace Init() with Reset()
    • 1c3dd38 : Android: Fix memory leak for remote MediaStream
    • 7391881 : Revert of Added send-thresholding and fixed text packet dumping. (patchset #4 id:160001 of https://codereview.webrtc.org/1266033005/ )
    • fdac516 : Disallow simulcast for H.264.
    • d828198 : Replaces SSLIdentity* with scoped_refptrRTCCertificate in the cricket::Transport layer.
    • d83df50 : Use RtcpPacket to send TMMBN in RtcpSender
    • c47a01d : Fix AppRTCDemo crash when room is connected after PC is destroyed.
    • 13d35f6 : Add check to prevent TURN usage if the protocol is not allowed.
    • 2f20fbe : Fix MIPS compile.
    • 0163fb2 : AudioCodingModuleImpl::Encode: Use a Buffer instead of a stack-allocated array
    • d838d27 : Added send-thresholding and fixed text packet dumping. Also a little squelch for the over-max-MTU log spam we see in there.
    • 3318f98 : VideoFrameBuffer: Make non-const data access explicit
    • 85ad62b : Remove per-frame captured frame logging.
    • af9fb21 : - Use C++11 loops in WebRtcVoiceMediaEngine/Channel. - Pull out part of WebRtcVoiceMediaChannel::SetRecvCodecs() into WebRtcVoiceMediaChannel::SetRecvCodecsInternal().
    • c464f50 : AndroidVideoCapturerJni: Fix threading issues
    • c464b40 : Android RendererCommon: Add unittests for getTextureMatrix()
    • 1eb87c7 : TCPConnection can never be deteted if they fail to connect.
    • 9b35115 : Move mock_nonlinear_beamformer to only be a header
    • b274547 : rtc::Bind: Capture scoped_refptr reference arguments by value
    • f4772ee : Get rid of unused types and constants in acm_common_defs.h
    • 1bb8cf8 : NetEq/ACM: Refactor how packet waiting times are calculated
    • 7230a21 : Android RendererCommon: Add unittests for getDisplaySize()
    • b6cac8f : Get rid of the manual destructor in AudioCodingModuleImpl
    • 87a8fbb : Fixing Pylint errors for plot_dynamics.py
    • 87713d0 : RTCCertificates added to RTCConfiguration, used by WebRtcSession/-DescriptionFactory.
    • dd00f11 : Remove no-op and unused methods from AudioCodingModule
    • 7ef9d91 : Android: Remove VideoRenderer.Callbacks.canApplyRotation()
    • bc2296d : Add a base class to Wav{Reader,Writer} to access shared parameters.
    • 41eeff4 : More iOS compile fixes.
    • deb4875 : Fix typos in https://codereview.webrtc.org/1230503003/ not caught by trybots.
    • dce40cf : Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t.
    • b594041 : TcpPort Reconnect should inform upper layer to start sending again.
    • 39b8eb3 : Fix Chromium GN build problem introduced in 608c3cfe
    • 4e14f09 : Add support for external decoders in ACM
    • e7cdc7f : No-op CL to test if video engine core problem solved.
    • d8ee4f9 : Use RtcpPacket to send BYE in RtcpSender
    • 608c3cf : iSAC: Make separate AudioEncoder and AudioDecoder objects
    • 2159b89 : Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
    • 9deaa86 : Fix initialization/termination of AudioDeviceTemplate
    • 7612f17 : Fix accidental redeclaration.
    • c0775c0 : Fix accessing uninitialized variables when not processing a reverse stream.
    • ea1012b : address comments from https://codereview.webrtc.org/1277263002/
    • 5bdafd4 : Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.""
    • 81a3e60 : Use RtcpPacket to send TMMBR in RtcpSender
    • dd4edc5 : Reland of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1300863002/ )
    • c232096 : Remove cricket::VideoProcessor and AddVideoProcessor() functionality
    • 9d15c66 : Include webrtc/base/json.h rather than from jsoncpp directly.
    • 22ff75a : Add unit tests for more packet types in rtcp_sender_unittest.
    • bfab5cb : Fix some minor errors with the voice engine caused by the refactor CL https://codereview.webrtc.org/1229283003/.
    • a5b273a : Fixing problems with RTP extension ID conflict resolution
    • 874ca3a : Don't do reconfiguration if recv codec order/preference changes
    • 5a3acd8 : First step of passive aggressive nomination. On the controlled side, a stun request without use-candidate attribute will be used for sending media.
    • fe3bc9d : Relanding "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied."
    • a1f590f : Add helper class GuardedAsyncInvoker to protect against thread dying
    • d3de9c5 : rtc::Bind: Capture method objects as scoped_refptr if they are ref counted
    • efefda6 : Move SystemInfo to rtc_base_approved and delete unused code.
    • ff020c0 : Android: Move common functions from VideoRendererGui to new RendererCommon file
    • 41b3a38 : Adds RTCCertificate, a reference counted object indirectly owning an SSLCertificate (by owning the SSLIdentity).
    • 9e260f1 : Prevent TimeUntilNextProcess log spam.
    • d476b95 : Android EglBase: Add helper functions to query the surface size
    • 081f34b : Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."
    • 3d564c1 : Add instrumentation to track the IceEndpointType.
    • 86cb923 : In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate.
    • 47872ec : In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate.
    • 5d69648 : Disabling TestUdpIPv6 on Linux
    • 048e80c : Revert of Revert "Remove CpuMonitor and related, unused, code." (patchset #1 id:1 of https://codereview.webrtc.org/1287913004/ )
    • c844ca4 : Move scoped_ptr.h to rtc_base_approved. This is more a formality than anything since scoped_ptr.h is already being included from rtc_base_approved code.
    • 1f4ffe0 : NetEq: Implement two UMA stats for delay adaptation.
    • a472e96 : Revert "Remove CpuMonitor and related, unused, code."
    • 370c884 : Revert "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied."
    • ba9ab4c : In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate.
    • 1a24012 : Remove CpuMonitor and related, unused, code.
    • 0a2955f : Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied.
    • bef77e2 : NetEq: Implement logging of Delayed Packet Outage Events
    • d84dcbd : rtpAnalyze matlab tool: filter out RTCP packets
    • 141c595 : Revert of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1290573004/ )
    • 35ab4ba : Use RtcpPacket to send REMB in RtcpSender
    • 7b3de4b : Re-enable LLVM LTO on Neon targets.
    • 3260133 : Fix -Wreorder compile error after https://codereview.webrtc.org/1189583002/
    • dbe5bd9 : Delete unused function SetSessionError.
    • b6d4ec4 : Support generation of EC keys using P256 curve and support ECDSA certs.
    • 1147702 : WebRTC Bug 4865
    • e930769 : Remove unnecessary neon flags and update a missing comment.
    • 805d8fb : Remove WebRtcIsac_Highpass_float().
    • 55e9a7d : Add Android VideoRendererGui events.
    • d332580 : Add stats overlay to iOS AppRTCDemo.
    • 60d9b33 : Integrate Intelligibility with APM
    • 03bb7c7 : Add LoudestFilter in ConferenceTransport
    • 4c530dc : Delete dummy dtlsidentityservice.[cc,h] files.
    • d5031fc : Android VideoRendererGui: Add dispose function
    • af5c035 : VideoCapturerAndroid: Release queued camera frames when stopCapture() is called
    • 38f8893 : WebRTC Bug 4865
    • ee8c6d3 : In PeerConnectionTestWrapper, put audio input on a separate thread.
    • 7437588 : Adding locking to webrtc::voe::Channel to fix race conditions
    • c558af8 : Removing DtlsIdentityService[Interface] which has been replaced by DtlsIdentityStore[Interface/Impl].
    • cf7f54d : Use RtcpPacket to send RPSI in RtcpSender
    • e2a8be1 : Revert of AppRTCDemo: Render each video in a separate SurfaceView (patchset #4 id:120001 of https://codereview.webrtc.org/1257043004/ )
    • d941b76 : Fix distortions of remote stream with odd size dimensions
    • 8a2cd3d : Revert H.264 HW encoder setting to CBR mode.
    • d6b243f : Enabling screensharing perf test. It should work now as the packet limit in the jitter buffer has been increased.
    • 05bfbe4 : AppRTCDemo: Render each video in a separate SurfaceView
    • fa30180 : Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
    • cc4ebad : Empty dtlsidentityservice.h/cc files added, to be removed once chromium gyp files don't reference it.
    • 5e56c59 : DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface).
    • 0365a27 : Use RtcpPacket to send SLI in RtcpSender
    • 4bc66fc : Fix data race in AMP.
    • 4de6622 : Fix a bug in computing audio delay on ios device. Converts seconds to milliseconds by multiplying 1000 instead of dividing 1000.
    • 3449faa : Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
    • 4cee419 : Separating voice activity flag from audio level in RtpHeaderExtension.
    • c2ee2c8 : Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
    • eb04d68 : Moved project configs to infra/config branch
    • 25c96d0 : Add thread checker to StatsCollection.
    • 2328a94 : Add average rtt to CallStatsObserver and an average rtt histogram.
    • 0482dcc : Enable HW H.264 decoding on Intel platforms.
    • 8381b37 : Removed bjornv from OWNERS and added two new owners
    • 2e1d8bb : Suppress a race in libjingle_peerconnection_unittest
    • fcf8ece : AndroidVideoCapturer: Return frames that have been dropped
    • c937139 : Regenerate bind.h using pump.py BUG=webrtc:4690 R=pthatcher@webrtc.org
    • a873644 : Move all the examples from the talk directory into the webrtc examples directory.
    • 5b4ce33 : DtlsIdentityStoreInterface added. New PeerConnectionFactoryInterface::CreatePeerConnection taking both service and store added (old CreatePC signature still exists).
    • 0c02264 : Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it.
    • bd10ee8 : Tiny cleanups.
    • 62dae19 : Use RtcpPacket to send FIR in RtcpSender
    • ef7228c : Selectable number of TL screenshare loopback test. Also contains some tweaks to make a single TL perform better.
    • 907dcfd : Increase packet limit in jitter buffer.
    • 37ec733 : VideoCapturerAndroid: Check if data is null in onPreviewFrame()
    • 0c85020 : Add list of devices with HW H.264 encoder non suitable for WebRTC.
    • 8d62971 : Fix race condition in EndToEndTest.AssignsTransportSequenceNumbers
    • b19eba3 : Fix Turn TCP port issue.
    • 867fb52 : Add support for transport wide sequence numbers
    • d67a219 : Switch to base/logging.h in neteq_impl.cc
    • 62cde2c : Disabling VP9 perf test
    • 503726c : Fix the generation mismatch assertion error.
    • 72aa9a6 : Use RtcpPacket to send PLI in RtcpSender
    • a9455ab : Integration of VP9 packetization.
    • 2386a45 : Supporting Pause/Resume, Sending Estimate logging. Corrected plot colors
    • a12ba55 : Added protection for GetCapabilities() failure.
    • 5f5f11c : FEC protect H264 delta frames as well.
    • 3641185 : Includes webrtc/build/protoc.gypi instead of build/protoc.gypi
    • b933667 : Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly."
    • 9a6e741 : Move audio_coding_module.gypi from main/acm2 to main/.
    • e2cb1f1 : Efficient Metric Recorder
    • 028cf48 : Added FullStack performance test for screensharing with VP9
    • c159b04 : Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly.
    • ee66016 : Added IsInBeam to mock_nonlinear_beamformer.h
    • d635895 : Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests.
    • 49c0ce3 : Revert "Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests."
    • 8993413 : Add a frame generator that allows scrolling over a larger still image, for use with new screen sharing quality tests. Also add support for this in the loopback tests.
    • a3b8769 : Add packetization and coding/decoding of feedback message format.
    • f1828e8 : Prevent OOB reads for truncated H264 STAP-A packets.
    • f38ea3c : Add support for VP9 packetization/depacketization.
    • 95b8718 : Fix to "Removing AudioMixerStatusReceiver and ParticipantStatistics"
    • 4540ffa : Removing AudioMixerStatusReceiver and ParticipantStatistics.
    • d40af69 : Split MoveReadPosition into Forward and Backward versions.
    • b3cc77f : Re-enable WebRtcIsacfix_AllpassFilter2FixDec16Neon
    • a446609 : When we trace to file, add eol of each trace message.
    • b3b79b6 : Clean up the Config to enable 48kHz support in AudioProcessing
    • ef35f06 : Remove webrtc::Config from ViEChannelGroup.
    • 081af25 : Remove kProtectionKey* and VCMKeyRequestMode.
    • fa37e33 : Add pbos@webrtc.org to webrtc/video_engine/OWNERS.
    • fe0c905 : Improve probing by ignoring small packets which otherwise break the mechanism.
    • b28678c : Add unittest to GlRectDrawer
    • 013a580 : VideoCapturerAndroid: Revert elapsedRealtimeNanos to elapsedRealtime
    • d55ce2d : BWE Simulation Framework: Standard plot logging
    • 7a1c24f : Remove "multichannel" from parameter to match interface name.
    • e2b34b7 : Bug fix: camera frames are dropped before wideo encoder.
    • 6bb1b6e : Control combined_audio_video_bwe with config bool.
    • cfd5f96 : Ignore packets with reordered timestamps when doing BWE.
    • a38233a : Removed extended jitter report from RtcpSender. This was never used (value always 0, when sent)
    • 6718e97 : Add encode and decode time to histograms stats: - "WebRTC.Video.EncodeTimeInMs" - "WebRTC.Video.DecodeTimeInMs"
    • c3f46a9 : iOS: Move AppRTC logging methods to public headers.
    • 28bae02 : Remove CircularFileStream / replace it with CallSessionFileRotatingStream.
    • 3ab2f14 : Remove C++11 calls from intelligibility_utils
    • 86c6d33 : Allow more than 2 input channels in AudioProcessing.
    • fcfdb08 : Update AUTHORS file.
    • d6fc47e : Remove base channel for video receivers.
    • 59adf34 : Evaluation test cases.
    • 66f438f : Revert of Fixing scenario where track is rejected and later un-rejected. (patchset #5 id:80001 of https://codereview.webrtc.org/1231613002/)
    • 64e753c : Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/)
    • b21fd94 : Temporarily disable ScreenshareSlides on Android.
    • c204754 : Allow more than 2 input channels in AudioProcessing.
    • 0b6a204 : Configure AudioProcessing directly in agc_harness.
    • b297c5a : Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .
    • 7c5304c : Allow webrtc compilation with stlport
    • 9341191 : Provides log sinks for rotating logs. Intended for use on mobile devices to record call logs.
    • f24b2bc : Modified histogram shell plot script, added python dynamics plot script
    • 235c35f : Implement store as an explicit atomic operation.
    • 085856c : Extend full stack tests with more stats
    • d89920b : Add resolution and fps stats to histograms: - "WebRTC.Video.InputWidthInPixels" - "WebRTC.Video.InputHeightInPixels" - "WebRTC.Video.SentWidthInPixels" - "WebRTC.Video.SentHeightInPixels" - "WebRTC.Video.ReceivedWidthInPixels" - "WebRTC.Video.ReceivedHeightInPixels" - "WebRTC.Video.RenderFramesPerSecond"
    • 65eb1c3 : Disable testcase NatTcpTest.TestConnectOut
    • d60a799 : Mark WebRTC project as public in luci-config
    • b69ab79 : VideoCapturerAndroid: Add function to change capture format while camera is running
    • 496019c : If the array size is even, the median should be average of its two middlemost elements.
    • 83d6b0c : Ignore genperf lib in merge_libs.py.
    • 343714e : Fix the problom that on Linux no external audio device can be selected since #9243.
    • 2981945 : Moved arrray_util include to beamformer.h
    • 8ff04d6 : Remove UpdateSsrcs from EncoderStateFeedback.
    • 324d9c9 : Avoids error message about unknown selected data source for Port iPhone Microphone
    • f421bdc : Fix an NPE when creating TurnPort with a NULL socket.
    • be37888 : Fixing scenario where track is rejected and later un-rejected.
    • b947f28 : Add pcap support to bwe tools. Allow filtering on SSRCs.
    • fabe2c9 : Remove deprecated functions.
    • c27d89f : Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame.
    • c5d0d95 : Ensuring that UDP TURN servers are always used as STUN servers.
    • d848d5e : Enable cropping window capturing for Win7 when Aero is disabled.
    • bd38428 : Don't use result of "field_trial::FindFullName" as string reference.
    • a9b4c32 : Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity.
    • 083b73f : Use std::string references instead of copying contents.
    • cd67022 : Define Stream base classes
    • cddb367 : Remove unused metric in overuse detector.
    • f393829 : Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used.
    • fb19f49 : Replaced uint32_t with standard uint16_t for sequence_number variables.
    • bf40b42 : Modified Simulation Framework Jitter Model.
    • 8fc7fa7 : Base A/V synchronization on sync_labels.
    • 9c261f2 : Supports logging for dynamic and histogram plots on Simulation Framework.
    • a4a8d4a : Base padding bitrate for an encoder on the bitrate allocated for that encoder, rather than the total bitrate of the channel group.
    • 3258db2 : Split iSAC encoder/decoder: Test more cases (and make sure they work)
    • 2d3b7e2 : AppRTCDemo file logging.
    • 43e7d3b : Avoid overflow in checking for emulation bytes in rbsp.
    • caa498a : Make sure RTCP is sent in tests when receiving packets even if REMB is delayed.
    • ba35d05 : Cleanup of iOS AudioDevice implementation
    • d6f1a38 : Remove ViEChannel simulcast lock.
    • 4988ca5 : Removed unused variables and the need to include the d3dx9.h file.
    • 870eee4 : Fix simulator issue where chokes didn't apply to non-congested packets.
    • a03cd3f : 1. Override and virtual has to be consistent. 2. provide an implementation for SetIceConnectionReceivingTimeout so that Chrome does not complain.
    • 6e2ce6e : Allow for framerate reduction for HW encoder.
    • 9009962 : Add methods to set the ICE connection receiving_timeout values.
    • 45d1fde : Revert of Fix simulator issue where chokes didn't apply to non-congested packets. (patchset #2 id:20001 of https://codereview.webrtc.org/1233853002/)
    • 662ae00 : Fix simulator issue where chokes didn't apply to non-congested packets.
    • 5d6e58e : Improvements to rtc::Bind
    • 30409b4 : Add statistics gathering for packet loss.
    • 35b72fb : Add new variance update option and unittests for intelligibility
    • d10a68e : Don't create unsignalled receive streams for RTX, RED RTX, and ULPFEC packets.
    • 8647922 : Revert the process noise co-variance of the bitrate over-use estimator to its value prior to r9545.
    • a6d2444 : Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code.
    • 4d9d097 : Fix follow-up in webrtc/test/field_trial.cc.
    • 97f44e1 : Remove a superfluous qualifier on an inline method.
    • 50cf10d : Make .gni less sensitive to type of arm_use_neon flags
    • 11324b9 : Wait for a longer time (5 seconds) before establishing the first bandwidth estimate.
    • bb36fdf : Remove empty-string comparisons.
    • 3b1e647 : Remove media sinks from Channel.
    • 0f620f4 : Make sure we process all pending offer/answer requests before terminating. This fixes a bug in the WebRtcSessionDescriptionFactory where messages would be dropped or worse yet processed after the factory was deleted.
    • 31acf3d : Add include_examples GYP variable.
    • e987a47 : Removed some unused variables in Windows code.
    • 6109386 : Expose the disable encryption option to JNI.
    • 5436051 : Add flakyness check based on the recently received packets.
    • aa97df4 : Roll chromium_revision 3ead4bc..f8d6ba9 (336983:337800)
    • cbd44e6 : Use Resampler default constructor in VAD
    • b8b0143 : Tighten link-local routing exclusion check
    • 6e89b25 : VP9 wrapper: Adjust speed setting.
    • d436298 : Remove ResetStatistics from RTP feedback.
    • 19492f1 : Add scoped class for overriding field trials.
    • a7d7054 : Remove VCM_*_PAYLOAD_TYPE constants.
    • c62642c : Make the BWE threshold adaptive.
    • 4e7aa43 : audio_processing: Adds two UMA histograms logging delay jumps in AEC
    • f935bcc : Use strcmp instead of == operator for c.name and name to find appropriate classes for WebRtcAudio*.java
    • 2bad88d : Prevent heap overflows for incorrect FEC packet lengths.
    • 468e62a : Remove MimdRateControl and factories for RemoteBitrateEstimor.
    • d92f267 : audio_processing: Changed kMinDiffDelayMs from 50 to 60 ms
    • 72a8cee : Targets should not depend on protobuf when enable_protobuf=0.
    • 894ad94 : Fix occurrences of const typed declaration without initialization
    • ac8869e : Report metrics about negotiated ciphers.
    • 366e952 : Follow-up: Remove old ReportedDelay AEC config
    • 2224294 : iSAC: Functions for importing and exporting bandwidth est. info
    • cd4a9bd : Remove decoder-thread instantiation for senders.
    • db0cf76 : Add test for dropping repeated NTP timestamps.
    • f4eca64 : iSAC: Pad with zeros instead of random data, to make testing easier
    • 0f133b9 : Rename APM Config ReportedDelay to DelayAgnostic
    • 0d7dbde : Update AppRTCDemo resolution for iPhone6/6+
    • a771bf8 : Fix some clang warnings with -Wmissing-braces in WebRTC.
    • d830aea : Add tkchin to video_coding OWNERS.
    • 0edd50c : Support for onbufferedamountlow
    • 545727e : Move early-return in TimeToSendPadding.
    • bd2522a : Fail RTP parsing on excessive padding length.
    • 8b80fb6 : Roll chromium_revision fbf756f..3ead4bc (336289:336983)
    • 4daa90e : Prevent size_t underflow in H264 SPS parsing.
    • 2f15093 : Prevent OOB read on truncated H264 headers.
    • 7ada923 : Prevent OOB reads for zero-length H264 payloads.
    • 48c3839 : Prevent depacketizer OOB reads on zero-length VP8 payload.
    • 6e355af : Added fields for configuration information to the protobuf format in the ACMDump. The ACMDump interface itself is not updated, so there is no way (yet) to actually write the configuration fields.
    • 2e43b26 : Prevent OOB reads in FEC packets without complete RED headers.
    • 1adbacb : Adding method IsInBeam to beamformer class.
    • 3c60d61 : Remove a cast again, after it was shown to worsen Windows perf.
    • 71f6f44 : iOS HW H264 support.
    • 70d5c47 : Prevent out-of-bounds reads for short FEC packets.
    • 1ca324f : Adds UMA histogram for system delay jumps
    • c689124 : Simplify OWNERS structure in modules/audio_coding
    • 9b9f338 : Adding Minyue to audio_coding/OWNERS
    • 9bc2c61 : Roll chromium_revision 9729297..fbf756f (335266:336289)
    • 241338e : Added support for keeping a buffer of the previous X seconds, to add to an AcmDump.
    • 4b91bd0 : Move frame input (ViECapturer) to webrtc/video/.
    • ebe7422 : Created SphericalPoint in array_util.h
    • 93fb53a : Adding a new ChangeLogger class to handle UMA logging of bitrates
    • ecf6b81 : Pull the Voice Activity Detector out from the AGC
    • 0ea42d3 : Send Sdes using RtcpPacket
    • 51c7cbb : Revert "Pull the Voice Activity Detector out from the AGC"
    • 518c683 : Pull the Voice Activity Detector out from the AGC
    • ac4234c : Add a [rtc_]build_with_neon variable to unify conditions.
    • 1c7075f : Ensure transient suppression is never enabled on mobile.
    • c0c3a86 : Prevent JS from bypassing RTP data channel bandwidth limitation.
    • 8d3e489 : Update deeper codereview.settings files to match the root.
    • 1b12cb0 : Enabling AudioDeviceTest.StartStopPlayout on Nexus 9
    • 59a677a : Android VideoRendererGui: Refactor GLES rendering
    • 2c4c914 : In screenshare mode, suppress VP8 bitrate overshoot and increase quality
    • 7ab5f80 : Adding an equals method for KeyValuePair for easier testing.
    • 66f920e : Remove definition of non-existent method. The private method "CreateDefaultLocalDescription" is defined in the class, but not implemented or used anywhere.
    • 084f387 : Reland mysterious cast that improves performance.
    • 6bfc82a : Test whether removing a cast still hurts performance.
    • 39b3100 : Change kEchoCancellation to be 'echoCancellation'. This is the second cl in WebRTC for this change and will be landed after Chromium has been updated to use kGooglEchoCancellation where that variant is required. See also the first part: https://codereview.webrtc.org/1179233003
    • 747d5f6 : Reland "Added ACM_dump protobuf, class for reading/writing and...", commit e9bdfd859c309991b4ea759587f39eecdbd42bd4. Changed the BUILD.gn file that was lacking some necessary items which caused Chromium to break. Original review: https://webrtc-codereview.appspot.com/52059005/
    • 97c9f8d : Remove iostream which causes a new static initializer TBR=pthatcher@webrtc.org BUG=webrtc:4576
    • 72cfd6c : Reland remaining bits of "Upconvert various types to int."
    • db4fecf : Attempt to reland: Allow intelligibility to compile in apm (https://codereview.webrtc.org/1182323005/)
    • be24c94 : Set / verify stats report timestamps.
    • 6b4a564 : Add UMA logging for target audio bitrate
    • bdc0b0d : Use RtcpPacket classes for SenderReport/ReceiveReport in RTCPSender
    • 9874ee0 : Add temporal-layers option to video_loopback.
    • ecb9a70 : Add AsyncInvoker files for chromium GN build
    • 6a688f5 : Add default downscale threshold to QualityScaler.
    • e8d191f : Restore rows() and cols() in aligned_array.h
    • 6ee4816 : Roll chromium_revision 6e6b751..9729297 (334984:335266)
    • 04465d2 : Revert of Fix PRESUBMIT.py after disabling CQ. (patchset #1 id:1 of https://codereview.webrtc.org/1192673003/)
    • 45fec22 : Revert "Temporarily disabling CQ due to infrastructure problems."
    • 7a75415 : Revert "Added ACM_dump protobuf, class for reading/writing and unittest."
    • 7f04b08 : Issue 4780: disabling multiple_routes breaks Turn/Tcp.
    • f260fc2 : Revert "Pull the Voice Activity Detector out from the AGC"
    • f5f8f52 : Revert "Increase the kMaxNoiseProbability in voice_activity_detector_test"
    • d4cec15 : Resolved Rebase Conflicts This is just https://webrtc-codereview.appspot.com/53629004/
    • 76eea37 : Workaround a (Windows) linker bug when doing a PGO build.
    • 39ffaea : Roll chromium_revision 441009c..6e6b751 (334775:334984)
    • c9b0f67 : Increase the kMaxNoiseProbability in voice_activity_detector_test
    • dc13abc : Initially when the design was to do this experiment in browser, which doesn't have webrtc code, it requires some glue code to bridge the difference between what's available in webrtc::base and browser process. Now since we're moving to renderer process, we could reuse a lot of existing interfaces instead of rolling our own.
    • 34be126 : Pull the Voice Activity Detector out from the AGC
    • ae37abb : Remove implicit-int-conversion warnings.
    • ff4ea93 : Only use paced packets for estimating bitrate probes.
    • 141725f : Fix PRESUBMIT.py after disabling CQ.
    • 3e89dbf : Add AudioEncoder::GetTargetBitrate
    • e9bdfd8 : Added ACM_dump protobuf, class for reading/writing and unittest.
    • 380884e : Temporarily disabling CQ due to infrastructure problems.
    • 7101269 : Reland "Revert "audio_processing/aec: make delay estimator aware of starving farend buffer""
    • 9d2fdac : Roll chromium_revision a08e761..441009c (334571:334775)
    • 2d627a6 : Add missing include guards for audio_ring_buffer.h. Yikes.
    • c555b99 : Revert of Allow intelligibility to compile in apm (patchset #1 id:1 of https://codereview.webrtc.org/1182323005/)
    • b7553df : Allow intelligibility to compile in apm
    • 01c9b01 : Revert "audio_processing/aec: make delay estimator aware of starving farend buffer"
    • 9002cc4 : audio_processing/aec: make delay estimator aware of starving farend buffer
    • 979e0b3 : Define uint64 and int64 using long long.
    • 6befa00 : Add presubmit trybot to CQ config.
    • 47cfc3a : Roll chromium_revision 4e76e79..a08e761 (334321:334571)
    • 986ee08 : Move default trybots configuration to CQ config.
    • f050b9d : Revert of Whitespace change (patchset #1 id:1 of https://codereview.webrtc.org/1182933006/)
    • 24b4eda : Add sent framerates to histogram stats: "WebRTC.Video.InputFramesPerSecond", "WebRTC.Video.SentFramesPerSecond".
    • 1d34fe9 : Adds support for webrtc::test::ResourcePath on iOS
    • b02af18 : Follow-up: Remove old DelayCorrection AEC config
    • 76381d9 : Update rtpAnalyze matlab tool to handle reordered packets
    • ac81163 : iSAC: Move global trig tables into the codec instance
    • 030249d : Initial SIE commit: migrating existing code
    • fe23090 : Whitespace change to test CQ
    • 524f784 : disable MacAsyncSocketTest::TestConnectFailIPv6
    • d10cd97 : Make global constants 'const'
    • 53dd4b1 : Roll chromium_revision c2239a8..4e76e79 (334133:334321)
    • a6aa6d9 : Fix a data race in AudioEncoderMutableImpl and derived classes
    • 05ce5dd : Roll chromium_revision e937e5f..c2239a8 (333350:334133)
    • 2b67925 : VideoCapturerAndroid: Add possibility to request a new resolution from the video adapter.
    • 70c7fe1 : Add kGoogEchoCancellation to MediaConstraintsInterface. This constraint will be equal to kEchoCancellation until we've updated Chromium to use kGoogEchoCancellation where that constraint is needed. Once that's done, I'll change kEchoCancellation to be 'echoCancellation'.
    • 01bbe3e : Fix AppRTCDemo crash under iOS armv7 devices
    • 782671f : Improve Android HW decoder error handling.
    • 2f65ac1 : Fix crash and warning in AppRTCDemo
    • 372f2fc : Connection resurrected with incorrect candidate type.
    • f564291 : Remove webrtc/libjingle/{examples,session}.
    • 36b7cc3 : Reland "Upconvert various types to int.", neteq portion.
    • bc440d5 : Revert "Reland "Upconvert various types to int.", common_audio portion."
    • 15b58ee : Reland "Upconvert various types to int.", common_audio portion.
    • bba7807 : Reland "Upconvert various types to int.", misc. codecs portion.
    • a8b335c : Reland "Upconvert various types to int.", ilbc portion.
    • aba07ef : Reland "Upconvert various types to int.", isac portion.
    • 7faba36 : Fix WebRTC window-capture to check for maximized state correctly.
    • 728d903 : Reformat existing code. There should be no functional effects.
    • b7e5054 : Match existing type usage better.
    • cb18097 : Revert "Upconvert various types to int."
    • 66a641a : Update encoder settings periodically, not only on new bandwidth estimate
    • 78fb3b3 : C++11 in-class member initialization in Call configs.
    • eb82309 : Remove FileMediaEngine.
    • 80cf97c : Android rendering: Move common EGL and GL functions to separate classes
    • f045e4d : Prepare to convert various types to size_t.
    • 786dbdc : Rename targets to use lower case format.
    • 9345e86 : audio_processing: Create now returns a pointer to the object
    • 8a19f3d : Relanding https://webrtc-codereview.appspot.com/56589004
    • 1fe120a : Add triggered checks.
    • a2c7940 : Ensures that modules_unittests runs on iOS
    • f4baca5 : Set mtu for DTLS to 1280
    • 2a10087 : Manual cleanups following clang-formatting.
    • 83ad33a : Upconvert various types to int.
    • 54b0ca5 : Revert "Landing https://webrtc-codereview.appspot.com/53669004/"
    • 2aef19c : Landing https://webrtc-codereview.appspot.com/53669004/
    • 532caea : Adding DCHECKs and constness to DtlsIdentityStore.
    • 441f634 : Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
    • 94a1232 : Roll chromium_revision b2c6a86..e937e5f (332773:333350)
    • 1b76ca1 : Auto-roll script: Add dirty tree check and git pull
    • ca84302 : Roll chromium_revision 3d86a83..b2c6a86 (332345:332773)
    • 8a89718 : Exclude Nexus 6 from OpenSL ES usage
    • 72e9f04 : Better determination of Symmetric NAT.
    • 0e1b229 : Disable TestGetAllPortsNoUdpSocketsNoTcpListen on valgrind memcheck.
    • 1b9add9 : Prevent bitrate overshoot for HD layer in VP8.
    • 61715ec : Fixed issue from previous commit, CL 56459004.
    • 04f4931 : VoE2 API draft
    • 77cabab : Enabling Packet-Loss plots for BweReceiver.
    • c1b9d4e : Add support for fragmentation in RtcpPacket.
    • 1aff095 : Moved check for native frame to VideoReceiveStream::FrameCallback.
    • 8f622a9 : Locking is no longer required with BoringSSL.
    • 55b6acb : Miscellaneous cleanups.
    • 349c2bb : Remove the timestamp_ member of StreamGenerator.
    • f291287 : Change "hybrid mode" to "|kNack| mode" in comments.
    • d7da120 : Disable reduced-size RTCP in default config.
    • fe55c38 : Removes automatic setting of COMM mode in WebRTC. It is now up to the application to ensure that it is in COMM mode before any audio streaming is started.
    • 3b2f67d : Disable some PortAllocatorTest on valgrind memcheck due to flakiness
    • eb66e80 : Re-land "Convert native handles to buffers before encoding."
    • 3fbf3f8 : Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
    • bdd185f : Added CQ config for WebRTC
    • 5f4b7e2 : Rename APM Config DelayCorrection to ExtendedFilter
    • efdce69 : Disable some PortAllocatorTest on asan due to flakiness
    • 7dbc076 : audio_processing/aec: Turn SignalBasedDelayCorrection to after 15 seconds
    • 85cf3c0 : Revert "Disable some PortAllocatorTest on tsan due to flakiness"
    • f019efa : Adding henrik.lundin to two WATCHLISTS
    • fc622cc : Move Requester to cc file.
    • 2a6b8b7 : Fix windows build break
    • d04d3d7 : Add SocketFactoryInterface::Prepare and fix how symmetric NAT is determined.
    • 26b0860 : Use one scoped_refptr.
    • 491bd53 : Disable some PortAllocatorTest on tsan due to flakiness
    • e973c2a : Remove win32toolhelp.h.
    • 59abdd9 : Whitespace change to test new Rietveld move.
    • 32130c6 : Move to Chromium's Rietveld at codereview.webrtc.org
    • b1825a4 : Change JitterBuffer::GetNackList to return a std::vectoruint16_t.
    • 248b0b0 : Run clang-format --style=Chromium on four files I'm otherwise touching.
    • a9952cd : Remove CHECK from GetThreadName. It's safe for prctl() to fail, so we fall back on noname for thread names if we can't get one, instead of crashing.
    • b4c5eaa : Fix a time control bug, that the VCMReceiver::FrameForDecoding may over sleep.
    • 73f7210 : Actively turns off platform-AEC when DA-AEC is used
    • 5abd3e1 : Revert r9359 "Implement NetEq's CurrentDelay function"
    • d8a03fa : Implement NetEq's CurrentDelay function
    • 60508f8 : Small changes to rtpAnalyze Matlab script
    • 6b99074 : Revert "Import org.junit.Assert instead of junit.framework.Assert."
    • a884709 : Import org.junit.Assert instead of junit.framework.Assert.
    • a398020 : SSL_set_read_ahead no longer needed with BoringSSL.
    • 308d163 : Revert "Convert native handles to buffers before encoding."
    • 14771ac : Fix Python lint and unit tests
    • 8f07418 : Roll chromium_revision 7779e7d..3d86a83 (332119:332345)
    • 84f81d8 : Fix implicit size_t to uint16_t warning on VS2015.
    • 8e6fd46 : Route time-stretching metrics through libjingle
    • 76cda01 : Document the time unit in EventWrapper.
    • 907bfb2 : Fix an apparent typo in a unittest that caused it to not actually check the new window list it fetched.
    • a831dc3 : Convert native handles to buffers before encoding.
    • 9ba52f8 : Remove intermediate RTCP CNAME buffers.
    • aff1c84 : Roll chromium_revision ccef3cb..7779e7d (331232:332119)
    • 5263b3c : Add options for NetEq fast accelerate mode through libjingle
    • 0908d0d : Fix issue with RTT computations in simulator.
    • 9b07368 : Revert "Roll chromium_revision ccef3cb..7779e7d (331232:332119)"
    • a8d686d : Roll chromium_revision ccef3cb..7779e7d (331232:332119)
    • f69f1fb : Testing and improving NADA algorithm.
    • 4765070 : Rename I420VideoFrame to VideoFrame.
    • c2cb266 : Match video orientation with device orientation for portrait and portrait upside down
    • 7be99bd : Revert "Match video orientation with device orientation for portrait and portrait upside down"
    • 14c2695 : Match video orientation with device orientation for portrait and portrait upside down
    • bc7dd7e : Add RTCConfiguration constructor to RTCPeerConnection wrapper.
    • d935f91 : Don't try to parse empty Ice urls.
    • a8202aa : Roll chromium_revision 1b9c098..ccef3cb (330302:331232)
    • 5a8bad6 : Update a comment that mentions the nonexistent Reset() method.
    • 5c6c6e0 : Implements TODOs for webrtc::datachannel state management when the SCTP association is congested. Adds missing state variables for each step in the transitions between DataChannelInterface::DataStates (kConnecting, kOpen, etc.), and uses them.
    • c28a896 : VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
    • bf738d7 : Temporarily disabling OpenSL ES for playout.
    • 04e5b49 : Make maximum SSL version configurable through PeerConnectionFactory::Options
    • cc84649 : Add LappedTransform accessors.
    • e70028e : Protect access to shared list of SRTP sessions.
    • 9e3cb33 : AppRTCDemo: check for necessary permissions before starting the call.
    • 770cc38 : Don't call CRYPTO_add in BoringSSL.
    • 3544837 : Disable reusing of ECDHE keys with NSS.
    • 5ee9f67 : Remove webrtcvideoengine.cc.
    • 603175a : Improve comments.
    • 7c4e745 : Support multiple URLs in PeerConnectionInterface::IceServer
    • 45b229c : Remove an unnecessary webrtc:: namespace prefix.
    • 92d9489 : Miscellaneous cleanups in VCMReceiver and its unit tests.
    • 645299d : Add frequency smoothing to postfilter.
    • d4f769d : Stop video candidates getting down to audio.
    • a743794 : audio_processing/aecm: Create() now returns a pointer to the object
    • 71861a0 : Remove GetSendSideDelay from RtpRtcp.
    • 7cd16b0 : video_processing_unittest: Only create files for visual inspection if the boolean flag 'gen_files' is set.
    • c3deaa3 : common_audio/vad: Removes head allocation failure check
    • 796e172 : Fixes crash in WebRtcAudioManager.setCommunicationMode
    • c41fe5d : Force 8 kHz sampling rate on Android emulator.
    • 2251d6e : Remove ViESender.
    • 259bd20 : Report ssrc_groups in GetStats().
    • 8bb6ea3 : Reset speech encoder before hooking it up to RED or CNG
    • 8051832 : Adding a new Matlab tool rtpAnalyze
    • 3b187b9 : Removed unnecessary includes of webrtcvideocapturer.h
    • 11beccd : Remove external report blocks from RtcpSender and rtp_rtcp interface.
    • 23c2e55 : Remove remaining .mk files.
    • b444b3f : Redirect logs to stderr in audioproc_f.
    • 9b720f7 : Add GetChunkLength to LappedTransform.
    • fec2c6d : Prevent potential double-free if srtp_create fails.
    • 1060260 : Added buildbucket bucket definitions
    • 92fbbb2 : Switch acm_receiver over to using base/logging.h
    • 9303eaf : Don't unnecessarily set mode/category on AVAudioSession.
    • def3988 : Configure default render delay as 10 ms.
    • cf808d2 : Add new fast mode for NetEq's Accelerate operation
    • cbe408a : WebRtcVideoCapturer: Getting rid of the |critical_section_stopping_| lock and all of its critical sections.
    • c065cc7 : Clarify boolean flags in neteq_opus_quality_test.
    • c13cacb : Remove an unused method in NetEq::Expand
    • de4703c : Refactor common_audio/vad: Create now returns the handle directly instead of an error code
    • afef4bf : Reland "Adding a test framework for conference mode application in VoE."
    • a4b7e5e : Revert "Adding a test framework for conference mode application in VoE."
    • 6a1ba8c : Fix coding style nits.
    • e87d487 : Fix ARM64 detection for VP8 and VP9 wrappers.
    • fc05205 : Adding a test framework for conference mode application in VoE.
    • 5d55c98 : WebRTC 4521: Remove usage of deprecated timezone global variable
    • 8d3ad82 : Script for auto-rolling chromium_revision in DEPS.
    • 5a3ebd7 : Revert "Remove default encoder/decoders."
    • e14e5f4 : Solve TSan warning about unlocking an unlocked mutex.
    • f09e09c : VoE: Remove unused interfaces
    • 32c2023 : Attempt at fixing error on the Chrome Windows FYI bots. It looks like our basictypes.h file in the overrides folder is including the file it is overriding due to include path precedence (Chrome's is lower than WebRTCs).
    • 905495c : Introduce NetEq::Config::ToString and use it in NetEq's constructor
    • e982a70 : PRESUBMIT: Fix typo.
    • 54be3e0 : Remove some WebRtcVideoEngine2 unittest stubs.
    • d8399e6 : Also provide sample rate when registering decoders
    • 323b132 : Protect ACM decoder buffer in stereo.
    • 57e5fd2 : PRESUBMIT: Improve PyLint check and add GN format check.
    • 00aac5a : Some cleanup for base/logging and base/stream.h
    • 23edcff : Move base/logging.* to rtc_base_approved.
    • ee369e4 : Refactoring of AudioTrackJni and AudioRecordJni using new JVM/JNI classes
    • a26c4e5 : Script to generate CL descriptions when rolling chromium_revision.
    • 0eefb4d : Detach base/logging.* from base/stream.*. This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl.
    • 469c2c0 : Make Config::default_value leak instead of having an exit-time destructor.
    • 4bf12ea : Revert "Fix sending wrong candidates down to transportchannel."
    • f65de84 : Fix sending wrong candidates down to transportchannel.
    • 67b635a : Fix simulcast_encoder_adapter giving full target_bitrate to the 2nd layer of any simulcast setup during InitEncode.
    • e4cb4e9 : Fix jitter buffer bug around out-of-order packets and non-RTX padding.
    • 4774874 : Enable AudioProcessing48kHzSupport by default
    • 3548dd2 : Set local SSRCs on receivers added before senders.
    • 367c868 : AudioEncoderCng: Handle case where speech encoder is reset
    • f761d10 : Update NetEq Quality Test.
    • 915df4f : CaptureManager: Don't stop a capturer at UnregisterVideoCapturer if it did not start in the first place.
    • 9a416bd : Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2
    • 5af6d47 : Code style change for quality_scaler.
    • 98d8cf5 : Hardware VP8 encoding: Use QP as metric for resize.
    • 5fdcdf6 : Enable ciphers to get ECDHE with NSS.
    • 6f2ef74 : Keep track of DTLS packet sizes to prevent partial reads.
    • a3ba0c7 : RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits
    • 36a1438 : Remove ViEFrameProviderBase.
    • af55ccc : Add RtcpMuxPolicy support to PeerConnection.
    • 02ff911 : Feature merge request: Add support for iOS http proxy detection
    • 523183b : Disables AudioDeviceTest.StartStopPlayout for Nexus 9 only
    • 280ed11 : Roll gtest-parallel.
    • 848d524 : Revert "RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits" https://webrtc-codereview.appspot.com/47229004/
    • 10022cd : RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits
    • 78ae00e : Remove default encoder/decoders.
    • b302ad4 : Remove unused VideoDecoder methods.
    • 1a07a1e : Solve data race in Pulse audio implementation.
    • 8602a3d : Cast to avoid char-interpretation of uint8_t in logs.
    • 05c7605 : Add resampling support in AudioBuffer::DeinterleaveFrom
    • 76b62ff : Clean up now-unused code that was used for libpeerconnection.[so|dll].
    • fce3242 : Remove linphonemediaengine.*
    • 8eb76ff : Make SHA1 computation thread-safe.
    • 5cdd702 : Add tools/vim to .gitignore.
    • 9b2b402 : Ensures that RECORD_AUDIO permission is required to start recording.
    • 5779d14 : Avoids crash when StartRecording conflicts with existing recording application
    • c3f4dbc : Remove rtp_rtcp/ dump functionality.
    • ca667db : Remove VCM debug recordings.
    • 831c558 : Allow setting maximum protocol version for SSL stream adapters.
    • 664cdaf : Replace assert() with static_assert() if the condition is evaluatable at compile time.
    • 5ca688b : Enable read-ahead on OpenSSL DTLS stream adapters. Prevent multiple BIO reads when reading header and body but read from internal OpenSSL buffer where possible.
    • 931e658 : Remove unnecessary dependencies for voe when building with include_internal_audio_device==0. In particular and practical terms, this avoids pulling in AudioDeviceModuleImpl and associated classes, in Chrome.
    • cb7f8ce : Clear ARM NEON flag
    • 4d71ede : Add HW fallback option to software encoding.
    • 97bce58 : Disable the EXPECT_DEATH check in bitbuffer on Android
    • bf560dd : remove filelock which is now unused
    • 17b889b : Issue 4366: Adapted frames have wrong width and height and are cropped.
    • 65de7d2 : Add a link to tools/vim to use the Chromium YCM config with webrtc.
    • 5ece00f : remove filelock which is now unused
    • 2f5be9a : Improve Android camera error handling.
    • 68898a2 : Remove AudioDeviceUtility.
    • df0c05b : Sort source file list for [rtc_]include_internal_audio_device. No code change.
    • c2b63fe : Adding Sony Xperia Z2 D6503 to HW AEC blacklist
    • 24e56e3 : Fixes Chromium FYI build issue on Android.
    • ccb49e7 : Remove Soundclip handling from libjingle.
    • 1ab67ae : Address the corner cases
    • b92be45 : Support 720P in portait as maximum on iOS.
    • 8db8069 : Change high frequency correction range
    • 3e95d3e : Don't log warning for unexpected STUN binding responses.
    • 79b2e06 : Make the BlockDifference() functions return DiffInfo as their callers expect.
    • 2e7a098 : Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc.
    • 7252a2b : Add HW fallback option to software decoding.
    • b261989 : Adding support for OpenSL ES output in native WebRTC
    • 02c9b36 : Roll gtest-parallel.
    • 7e0c7d4 : Add support for external encoders in ACM
    • ea14f0a : Move SetCurrentThreadName to platform_thread.* in rtc_base_approved, update all webrtc and libjingle code to use the same function and remove extra implementations.
    • bd1bc47 : Restructure decoder registration in ACM
    • 9d8b71e : Remove some dead code in ViEChannel.
    • a6e883b : Fix constant in SetCurrentThreadName.
    • bebc690 : Add platform_thread source files and move types from thread_checker_impl to there.
    • 24ec128 : Roll chromium_revision 5118a5b..1b9c098 (330060:330302)
    • a7d03ae : Roll chromium_revision 62a5bb3..5118a5b (329063:330060)
    • 144d018 : fix indent on tokenize_first function signatures
    • 42af6ca : Add logging of "use candidate" and when we switch ICE "best" connections.
    • b2d2623 : Don't use rtc::LogCheckLevel, because it breaks Chrome.
    • 1cf6f81 : Add logging for sending and receiving STUN binding requests and TURN requests and responses.
    • 37931c4 : Stunprober interface, its implementation and a command line driver.
    • 0e07f92 : Split fmtp on semicolons not spaces as per RFC6871
    • 20f3f94 : Clear bitrate stats for unused SSRCs.
    • 4cd6940 : Enable -Wformat-security warning and cleanup GYP.
    • 39f2b0c : Implemented video device info for iOS
    • a4463b2 : Further updates to fix libjingle logging. Since libjingle log constant values decrease as severety goes up while Chrome's increase, I decided to handle the verbosity level check explicitly and convert libjingle severity over to chrome constants only when we log. This also requires updating the unittests on the Chrome side.
    • 99eeee3 : Fix logging in Chrome. The constants we were using for severities don't match Chrome's, so I added a little translation function. A longer term fix could be to simply use the same values as in Chrome to not need the translation. That will however be a bigger change.
    • 06c577f : Set msvs_error_on_missing_sources=1 in GYP_GENERATOR_FLAGS on Windows.
    • 2013aec : Propagating RTT from send-only channel to receive-only channel.
    • 0703766 : Fix issue where receive-side encoders are included in the padding bitrate.
    • 9a63866 : Move IncomingVideoFrames to common_video/.
    • 4feb505 : Remove VideoProcessing::ColorEnhancement.
    • 5ec9985 : Windows utility to setTheadName to help debugging.
    • 9b9f1c4 : Remove basictypes.h dependency from bitbuffer.
    • e235714 : Guard new protobuf target with enable_protobuf==1.
    • 300eeb6 : Remove VideoEngine interfaces.
    • 8171735 : Add NetEqIlbcQualityTest
    • df66453 : Remove FPS-kilo-FPS conversion in VideoSender.
    • e5ff00a : Add NetEqPcmuQualityTest
    • fade179 : Remove leaking aecdump testfiles.
    • 075bb8d : Fix race in AudioCodingModuleImpl::Add10MsData()
    • 1b794d5 : Switch to use SHA-256 for certificates / fingerprints.
    • cb3e8fe : Increase the tolerance in NetEq's DelayManagerTest a notch
    • 67c9df7 : Base NACK on send codecs.
    • 126c03e : Base decision to send REMB on send codecs.
    • 64dad83 : Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
    • 092041c : Setting OPUS_SIGNAL_VOICE when enable DTX.
    • 9f7908e : Roll chromium_revision ec5b768..62a5bb3 (328242:329063)
    • 242e22b : Refactor RTCP sender
    • 1f62923 : Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
    • fd32f35 : Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
    • 54adb28 : mac: Explicitly redeclare methods only available on 10.7+.
    • 4c277bb : Add basic SCTP packet logging.
    • cdb47a4 : Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
    • 45553ae : Remove VideoEngine interface usage from new API.
    • 208a229 : Adding a new constraint to set NetEq buffer capacity from peerconnection
    • 83b5c05 : Modify NetEqQualityTest
    • cb05b72 : Add WAV and arbitrary geometry support to nlbf test.
    • d3ddc1b : Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.
    • e444a3d : WebRtcVoiceEngine: Get rid of unnecessary template base class.
    • aaf8ff2 : WebRtcVoiceEngine: virtual to override + git cl format.
    • 6179b89 : Remove unused API on WebRtcVoiceEngine.
    • 2ea71c3 : Replace ACMGenericCodec with CodecOwner and AudioEncoderMutable
    • 53d0dc3 : Wire up RTT to send-side GCC and TCP.
    • 4b60c73 : Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
    • dcccab3 : New interface: AudioEncoderMutable
    • 81ea54e : Remove WebRtcVideoEngine.
    • ccfc939 : Reinterpret AudioOption delay_agnostic_aec to override HW-AEC
    • c81591d : NADA's proposal from Cisco.
    • f353dd5 : VoE: cleanup VoENetwork implementation
    • 1ff218f : audio_processing/aec: Do not scale target delay at startup when on Android
    • 532531b : audio_processing/delay_estimator: Always update robust validation statistics
    • 40a6d59 : audio_processing/tests: Adds a flag to unpack input data to text file
    • 9695d85 : Added VP9FrameBufferPool, a memory pool that is shared between libvpx and webrtc. Using the VP9 codec, the libvpx decoder will obtain its buffers from our memory pool. This lets us reuse the same buffers for our I420VideoFrames and not have to copy a frame for every decode (from libvpx buffers to webrtc/I420VideoFrame buffers).
    • f242e66 : Replace asm NEON function by intrinsics implementation on ARMv7
    • 507a550 : Delete auto-roll script since moved into Chromium.
    • 589699e : Fix bug in transform_neon.c in iSAC codec.
    • 57cc74e : iOS camera switching video capturer.
    • 5cb9ce4 : Remove ViECodec usage in VideoSendStream.
    • ab00404 : VCMEncodedFrame::VerifyAndAllocate: Use size_t instead of uint32_t for size argument
    • 01b4888 : Use padding to achieve bitrate probing if the initial key frame has too few packets.
    • 78c8bbf : Roll chromium_revision 0cb2549..ec5b768 (327252:328242)
    • c56ac1e : rtc::Buffer: Remove backwards compatibility band-aids
    • f75f0cf : Enable GoogleWifiTrace3Mbps simulations.
    • 0d26605 : VoE: apply new style guide on VoE interfaces and their implementations
    • 79c1433 : Delete VoiceChannelTransport before deleting Channel in voe_cmd_test
    • 0b15445 : VoE: Follow-up to https://webrtc-codereview.appspot.com/49759004/
    • e433c0e : Restore back verbosity logging for camera captured frame.
    • f2f8283 : Use rtc::CriticalSection in webrtc/video/.
    • cac1b38 : Expose RTCConfiguration to java JNI and add an option to disable TCP
    • 4eddf18 : Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle.
    • 8a6680e : Remove base/move.h (no one uses it anymore)
    • cbf0927 : Revert "rtc::Buffer: Remove backwards compatibility band-aids"
    • 9e1a6d7 : rtc::Buffer: Remove backwards compatibility band-aids
    • ff019b0 : Move rtc::AtomicOps to webrtc/base/atomicops.h.
    • f16fcbe : Remove ViECapture usage in VideoSendStream.
    • 46bd31b : VoE: VoENetwork unit test
    • 3cfa756 : audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz
    • efbde37 : Don't use CPU adaptation for screen content in the new API.
    • adf89b7 : Added SetBitRate function to VoE API to allow changing the audio bitrate.
    • 23fba1f : Add AudioReceiveStream to Call API.
    • 10ba3ee : Roll chromium_revision a12e1e1..0cb2549 (326495:327252)
    • dea11f9 : Add per flow throughput and delay metrics.
    • 94cc1fe : Remove ViEImageProcess usage in VideoSendStream.
    • c444de6 : Make setup_links.py handle non-link directories during cleanup
    • 1ba344a : Adds a MediaConstraint for the AudioOption aec_dump
    • 97f13c5 : Fixed incorrect RBSP parsing. The original version would eat 0x3 as an emulation byte in places where it shouldn't, whereas the real parsing is only supposed to eat 0x3 preceded by 0x0 0x0.
    • 86153c2 : Added a BitBufferWriter subclass that contains methods for writing bit and byte-sized data, along with exponential golomb encoded data.
    • 80154f6 : Set correct .type directive for asm functions.
    • faa6d07 : Remove a few verbose log messages from webrtcvideoengine2.
    • 019087f : Add safeguards against signalling peer-reflexive candidates.
    • ae33134 : Always specify current OS when syncing Chromium.
    • 8786f63 : Roll gtest-parallel.
    • 31dc737 : Platform dependent way of generating the seed for srand for simulations, so that they can be run in parallel.
    • 88de479 : AudioEncoderIsac: Print error code if CHECK for successful encoding fails
    • bcbcd84 : Improve TCP implementation by adding ssthresh and make it possible to start it with an offset.
    • 9d657cf : Fix dangling pointer in screenshare_loopback
    • beb9798 : audio_processing: Fixed incorrect usage of SetExtraOptions() in offline tool
    • ddbddbd : Remove ViENetwork usage in VideoSendStream.
    • 038df3c : Remove ViEExternalCodec usage in VideoSendStream.
    • 4a9cb6b : Prevent zero-timestamps in captured_frame_.
    • 143cec1 : Set correct encoder-specific settings for vpx in the new API.
    • e8a197b : Enable isac NEON building on Aarch64
    • d7e5c44 : STUN allocation should not be disabled when using shared port and TURN servers are provided.
    • 5a92aa8 : Add 3-band filter-bank implementation
    • 494f209 : Move CriticalSection into rtc_base_approved.
    • 59d91dc : Remove ViERTP_RTCP usage in VideoSendStream.
    • e6cefb6 : GYP variables for building expat, icu, libsrtp, usrsctp
    • 61be2a4 : Clean up RTCPSender.
    • 3c391cb : Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api.
    • 52ef9d7 : Stop IncomingVideoStream on delete.
    • 23dc68e : Add the rtc_build_openmax_dl variable to the GN build.
    • 12e0329 : Do not use Magnifier if there are multiple screens since it sometimes crashes.
    • 77d444a : Handle the case when hoststring is empty.
    • c4188fd : Use IncomingVideoStream in VideoReceiveStream.
    • f955b5d : Add h.264 AVC SPS parsing for resolution (re-land)
    • c043afc : Cleanup inside IncomingVideoStream.
    • a9ae0df : Roll chromium_revision d5098d0..a12e1e1 (326014:326495)
    • a96f02b : Make sure histograms in jitter buffer are only updated if running.
    • affcfb2 : Refactor common_audio/signal_processing: Removed usage of trivial macro WEBRTC_SPL_MUL_16_16
    • e3827f2 : Revert "Add h.264 AVC SPS parsing for resolution."
    • 5ea8eff : Add h.264 AVC SPS parsing for resolution.
    • 9728241 : Record H264 NALU type in the h264 header.
    • fe7a80c : Prevent sender RTCP signals for receive-only channels.
    • 7f287cc : rtc::CriticalSection: Add dummy implementation of IsLocked for release builds
    • 24d4485 : Enable -Wunused-private-field warning for talk/
    • d3e8eda : (Re-land) AudioEncoderDecoderIsac: Merge the two config structs
    • 92f9eac : g722 and red encoders: Use rtc::Buffer instead of scoped_ptruint8_t[]
    • 261f644 : Suppressing VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Dr.Memory
    • 6bf1084 : rtc::CriticalSection: Add function IsLocked
    • bd67f66 : Restore webrtc/base/move.h, because it's used in Windows Chromium builds
    • 3525954 : Use short include paths for icu headers.
    • 915590e : Moved ByteBuffer/BitBuffer into rtc_base_approved.
    • 01aeaee : Fix GetSignatureDigestAlgorithm for openssl to prepare for EC key switch.
    • a8e285d : Remove webrtc/base/move.h, and make types move-only manually
    • ee0b00e : Prevent recv-stream reconfig on identical codecs.
    • 908e77b : Allow Java code to detect if VP8 and H.264 HW decoding is supported.
    • b672882 : Move cricket::FakeCall and associates to a separate file.
    • 7fb711f : Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class.
    • 96d1d89 : Do not register bandwidth observer for receive only channels. An incoming rtcp report block is inserted to both send and receive channels in Call::DeliverRtcp. The report block may also be accepted by each receive channel (in addition to the send channel) but fails to calculate the rtt (=0). Remove registration of bandwidth observer for receive channels. Prevents multiple callbacks to the bitrate controller (and with incorrect rtt) for an incoming report block.
    • 393347f : Report receive-side packet loss.
    • 7c027b6 : Enable more Clang warnings for talk/
    • 5a31780 : Reformatting RTPtimeshift.cc file.
    • ac69016 : Improve TCP by adding a real timeout to in flight packets.
    • 8e4b9e8 : Roll chromium_revision dcb0929..d5098d0 (325030:326014)
    • e555b7b : Fix CC flags in GN Windows build.
    • fb49451 : Disables mic bump-up level if not built with chromium
    • 8f85dbc : Reduce the number of registers used in MIPS optimizations.
    • bbf7c86 : Add a new BitBuffer class to webrtc base.
    • 61b4d51 : Dynamic resolution change for VP8 HW encode.
    • 5464a6e : Remove VideoCodingModule::InitializeReceiver.
    • 9dbbcfb : Remove VideoCodingModule::InitializeSender.
    • 9570224 : Fix broken perf prints.
    • 5f92051 : Fix bug in TCP implementation (simulations).
    • e62202f : Support handling multiple RTX but only generate SDP with RTX associated with VP8.
    • 6cff9cf : Revert "Remove simulcast modules from ViEReceiver."
    • 06b08af : VoE: VoEBase unit test
    • c4905fb : Fix race condition in Android camera JNI code.
    • ac7d97f : Remove frame copy in RTCOpenGLVideoRenderer.
    • 011c00f : rtc::Buffer: Accept void* in addition to the byte-sized types
    • 8c05415 : Add extra logging for Android camera JNI layer.
    • 9478437 : rtc::Buffer improvements
    • 9154373 : Do not define POSIX.
    • d43ba89 : PRESUBMIT: Add new trybots for iOS Simulator
    • 599beb8 : Revert "AudioEncoderDecoderIsac: Merge the two config structs"
    • 09a9ea8 : Supporting formats of non-multiple of 16 widths on Android.
    • a51e8f4 : Fix some simulation issues.
    • 14a97f0 : Remove simulcast modules from ViEReceiver.
    • 1d19893 : Add TCP fairness test.
    • b0b5425 : Let rtp_analyze parse absolute sender time
    • 61c2a6f : Remove rtc::Buffer::length(), since no one uses it anymore
    • d4e8014 : Fix build errors in r9022 / 09bdc1e5f5a9.
    • 09bdc1e : Add a BWE fairness test.
    • 3795937 : Adds a simplified Reno-type TCP sender.
    • f49dbfa : Close all camera resources when camera error happens.
    • 3f4eed0 : Deliver RTCP packets only once per receive stream.
    • fb98c40 : Register RTP/RTCP modules outside rtp_rtcp_cs_.
    • 382c58d : Move target_subarch from gyp_webrtc to supplement.gypi
    • f2497cf : Fix unknown option '-msse2' warning
    • 7c324ca : AudioEncoderDecoderIsac: Merge the two config structs
    • 9829af4 : Disable VP8 encoder HW acceleration on NVidia devices.
    • 7d89f80 : Use BoringSSL as default on iOS
    • 5d22c00 : Add performance tests flag to audioproc_float
    • 41ee1ea : Modified the simulcast encoder adapter to correctly handle encoded frames from sub encoders even if the encoder is unable to (temporarily or permanently) produce frames of the exactly matching resolution. This is done by using a different EncodedImageCallback for each encoder, which remembers which VideoEncoder it is registered to and forwards that on to SimulcastEncoderAdapter::Encoded.
    • 099323e : Have ViE sender also use the last encoded frame timestamp when determining if the video stream is paused/muted, for purposes of padding.
    • 352b2d7 : Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream).
    • 4b76c02 : Roll chromium_revision 8af41b3..dcb0929 (324854:325030)
    • 3c3f646 : Prevent null-stream reconfigs on RTP extensions.
    • 36fc1ba : Update renamed Android ARM64 trybot in PRESUBMIT.py.
    • c317ce5 : VoE: move mock directory 1 level up
    • adc46c4 : audio_processing/agc: Adds config to set minimum microphone volume at startup
    • 19a3807 : Updated .gitignore to ignore isolate_deps_dir.
    • a9c0ae2 : Add a sparse FIR filter implementation
    • e432800 : Enable CPU adaptation by default.
    • fcf54bd : Reland "Avoid critsect for protection- and qm setting callbacks in VideoSender."
    • 73ba7a6 : Remove PORTALLOCATOR_ENABLE_BUNDLE, PortAllocatorSessionProxy, PortAllocatorSessionMuxer, and PortProxy.
    • 74b9769 : Deliver RTCP packets only once per send stream.
    • 2dd6a27 : VoE: format VoEBase according to new style guide
    • 0de7bcf : Removes use of AudioManager.setSpeakerphoneOn in audio manager
    • 6739952 : Roll chromium_revision 70a0480..ac81bcc (324430:324836)
    • 56d5028 : Remove SignalCaptureStateChange from MediaEngine.
    • 575a802 : Add an option to update mirror flag in Android video renderer.
    • 1b67795 : Add i386 to ios fat library build script and use boringssl.
    • 529921e : Explicitly set target_subarch for iOS on ia32/x64
    • 77f0e3f : Remove GetStartCaptureFormat and some related code.
    • 6ae2572 : Add missing configuration of rtx payload type for rtp/rtcp module.
    • 03dec77 : Add chromium/_bad_scm to .gitignore
    • 0f911d7 : Refactor audio_processing/nsx: Removed usage of macro WEBRTC_SPL_MEMCPY_W16
    • 61a4b04 : Refactor common_audio/vad: Removed usage of trivial macro WEBRTC_SPL_MUL_16_16(a, b)
    • e7b221f : Remove deadlock in WebRtcVideoEngine2.
    • 6fc2d2f : VoE: revert CHECKs into asserts
    • 9e5e421 : VoE: cleanup VoEBaseImpl
    • 93ef1d8 : Change ACM's CodecManager to hold one encoder instead of an array
    • eba964f : Revert "Support none multiple of 16 pixels width on android."
    • 99c2fe5 : Fix NullVideoEngine's CreateChannel implementation.
    • b32a5c4 : Add more logging around TURN refreshes.
    • e4ae8d8 : Changes in VideoCapturerAndroid.
    • f4acf46 : Support none multiple of 16 pixels width on android.
    • 3949e86 : Prevent decoder busy loop for send-only channels.
    • a125d7d : Changes default audio mode in AppRTCDemo to MODE_RINGTONE. Also prevents that we try to restore audio mode when it has not been changed.
    • e12a667 : Remove i420_video_frame.h from common_video.gyp
    • 9bfe3da : Cleanup: Remove i420_video_frame.h header.
    • f6c003e : cricket::VideoFrameFactory: Handle if created frame is null
    • 9526187 : Default enable abs send time bwe for CallTest
    • 09bf1a1 : Delays changing to COMMUNICATION mode until streaming starts. Restores stored audio mode when all streaming stops.
    • 0184057 : VideoAdapterTest: Replace FileVideoCapturer with FakeVideoCapturer
    • dcbd3ac : Improve BWE plotting and logging to make it possible to use multiple windows/figures.
    • f2822ed : Refactor audio_coding/codecs/isac/fix: Removed usage of macro WEBRTC_SPL_MUL_16_16_RSFT
    • f6a99e6 : Refactor audio_processing: Free functions return void
    • 0666a9b : Remove Transport::Reset, which is never used, and only makes reading the code harder.
    • f9bbbdd : Roll chromium_revision d8f8dc8..70a0480 (324211:324430)
    • d417c93 : Remove android_webview_build conditions.
    • 9504b89 : Cleanup: Remove unnecessary SHA1Transform() declaration.
    • 3a93986 : Exit after printing usage message.
    • 7f6c4d4 : Fix clang style warnings in webrtc/modules/audio_coding/neteq
    • 4117775 : Roll chromium_revision 5333e14..d8f8dc8 (323410:324211)
    • 76c53d3 : Remove ViE interface usage from VideoReceiveStream.
    • 15cf019 : Add field-trial flag to disable WebRtcVideoEngine2.
    • 9b3f56e : Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection."" This reverts commit e41d774c4d0a60066866fc2d0ae48dd0e839ff23.
    • 2c37078 : Fix crash with CVO turned on for VP9 codec
    • d61ebda : Fix the sigslot type of DtlsIdentityStore::WorkerTask.
    • 036b420 : Updated iOS video capturer to take device orientation into consideration.
    • 1064679 : Revert "Fix crash with CVO turned on for VP9 codec"
    • 29b1a1c : Fix crash with CVO turned on for VP9 codec
    • c78da7d : Roll chromium_revision 719b839..5333e14 (322539:323410)
    • 06c8013 : Explicitly set target_subarch for iOS (re-land)
    • fbfc74a : Increase filename size for dummy device factory.
    • ad1f9b6 : Remove warning on input frames before config.
    • 64c0366 : Revert "Revert "Split EventWrapper in twain.""
    • ca047f7 : Stop building NSS on Windows.
    • 968b0e2 : Removed build dependency on er_tables_xor.h, since it has been deleted
    • 2e266e9 : On iOS, detect pdp_ip networks as cellular.
    • fbe5b31 : Fix merge_libs.py to correctly use the IGNORE_PATTERNS list.
    • 9e420af : Fix potential race conditions in Android video renderer.
    • e41d774 : Revert "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection."
    • 3b71efc : Revert "Explicitly set target_subarch for iOS."
    • 2519c45 : Fix clang style warnings in webrtc/modules/audio_coding
    • f6b7265 : Explicitly set target_subarch for iOS.
    • 1d83f1e : talk/media/webrtc/webrtcvoiceengine: Delay Agnostic AEC should not override HW-AEC
    • 49a862e : Return pending buffers to Java VideoCapturerAndroid if camera is stopping BUG=4510 R=magjed@webrtc.org
    • 26679d6 : ViEFrameCallback::DeliverFrame: Make I420VideoFrame const ref.
    • 3211934 : Fix build breakage in WrappedI420Buffer::native_handle() Sorry... My cl broke the build since I had not properly rebased and tested. https://webrtc-codereview.appspot.com/43999004/ TBR=mflodman@webrtc.org
    • 75db861 : Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.
    • e1c1ee2 : EncodedVideoData is unused, so remove it
    • e095148 : Port some fixes in AppRTCDemo.
    • be508a1 : Implement Tcp Reconnect for TCPPort.
    • ef88309 : Cleanup: Forward declare AudioFrame type in voiceprocess.h
    • ae0f0ee : Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
    • 7351f46 : Don't send STUN pings if we don't have a remote ufrag and pwd.
    • bc4b934 : Add a DCHECK to RegisterModule to make sure it's called on the controller thread.
    • 7f375f0 : ProcessThreadImpl - hold the lock while checking thread_ and calling ProcessThreadAttached(). This is needed since DeRegisterModule is currently being called on arbitrary threads.
    • 3354419 : Zero copy AndroidVideeCapturer. This cl uses the YV12 buffers from Java without a copy if no rotation is needed. Buffers are returned to the camera when the encoder and renderers no longer needs them.
    • 037bad7 : ~CaptureManager: DCHECK(capture_states_.empty()) instead of CHECK until we fix not empty bug.
    • cb76b89 : Cleanup: Move json.h into rtc namespace.
    • 0dd5802 : Update callers to include messagedigest.h.
    • db313b6 : Disable EndToEndTest.ReceivedFecPacketsNotNacked on all platforms.
    • d4e7501 : Refactor audio_coding/codecs/isac/fix: Removed usage of trivial macro WEBRTC_SPL_LSHIFT_W32()
    • 64c1e8c : Enable CVO by default through webrtc pipeline.
    • aaf61e4 : Cleanup: Remove MD5_CTX typedef.
    • fa16dda : Revert "Port frame_analyzer and rgba_to_i420_converter targets to GN build."
    • 6ac53b2 : Port frame_analyzer and rgba_to_i420_converter targets to GN build.
    • 722ef1f : Remove henrike@ from OWNERS
    • cf3c83e : Revert "Split EventWrapper in twain."
    • 31331cf : Revert "Enable CVO by default through webrtc pipeline."
    • d91cb5d : Reduce the number of Chromium dependencies synced.
    • 3cd9eaf : Ensures that AudioManager.isVolumeFixed() is only used for Android L and above
    • f536a50 : Remove duplicated source listing of gtest_prod_util.h
    • f809b9b : Fix bug in WebRtcIsacfix_FilterMaLoopNeon.
    • 9cb1f30 : Remove er_tables_xor.h.
    • 1b1c15c : Enable CVO by default through webrtc pipeline.
    • 4b3c0d6 : Use WebRTC API to convert byteorder in srtpfilter.
    • 4825356 : RTCDataChannel: Unregister data channel observer on dealloc.
    • 379069f : VideoRenderCallback::RenderFrame: Make I420VideoFrame& ref const.
    • 0828a0c : Revert "Avoid critsect for protection- and qm setting callbacks in VideoSender."
    • 23914fe : Reject RTP one-byte extension ID 0.
    • 903c0f2 : Avoid critsect for protection- and qm setting callbacks in VideoSender.
    • 738a5b4 : Remove old suppression for ProcessThreadImpl. The implementation has been changed considerably since it was added.
    • bc46bf2 : common_audio: Explicit cast in WebRtcSpl_NormW16 on ARM
    • 0194d32 : Add WebRtcAudioManager to peerconnection_jar library
    • 65f74a1 : Revert "Suppress data races in libjingle_peerconnection_unittest"
    • 2c9c83d : Remove non-functional asynchronous resampling mode.
    • 45c6449 : Introduce CodecManager and move code from AudioCodingModuleImpl
    • f7b9cf5 : Suppress "EndToEndTest::ReceivedFecPacketsNotNacked" on Asan, Tsan
    • 842a4a6 : Add locks to Start(), Stop() methods in ProcessThread. This is necessary unfortunately since there are a few places where DeRegisterModule does not reliably occur on the same thread.
    • 22e209d : Introduce AudioCodingModuleImpl::current_encoder_
    • 582f80e : Clamp decoder sample rate to 32000 in iSAC
    • 1ecfd55 : videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)'
    • 451b614 : Fix gyp path for bwe simulator include.
    • 8e9c67e : Suppress data races in libjingle_peerconnection_unittest
    • 9f52448 : Roll chromium_revision 4d63ee8..719b839 (322012:322539)
    • 6b3ccfc : GN: Cleanup no longer needed libvpx config.
    • 819011c : Additional suppression for TSan deadlock detection
    • dfd53fe : Raise streams for SetMaxSendBitrates above 2000k.
    • 53eda3d : Add tests for r8811.
    • b3fc48b : Update the notice about the slow Chromium sync.
    • 1d36003 : Suppress TSan errors triggered when deadlock detection is enabled.
    • 9ff73f5 : Final minor fix in WebRtcAudioManager
    • 424694c : audio_processing/agc: Put entire method set_output_will_be_muted() under lock
    • 75a0255 : Handle borked Android cameras gracefully. It turns out that Camera.getCameraInfo can throw an exception if the camera does not work.
    • 8324b52 : Adding playout volume control to WebRtcAudioTrack.java.
    • 8ed6a4b : Remove unused non-standard capture stats.
    • 3954e1d : Remove unused implementations in cricket::VideoFrame
    • 7100dcd : Adding "usedtx" as Opus codec parameter.
    • bef8d2d : Add a lock to NSSContext to fix data race
    • b8cfa68 : Update speed setting in VP9.
    • 74d9ed7 : Report send codec name in GetStats().
    • d6f4c25 : Reject streams reusing simulcast or RTX SSRCs.
    • a990784 : AcmReceiver: index decoders by payload type instead of ACM codec ID
    • 9b5f96e : Add some sanity CHECKs to webrtc::Call.
    • c79f7ed : Fix build error introduced by r8864.
    • e590416 : Moving the pacer and the pacer thread to ChannelGroup.
    • 5225dd8 : If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size.
    • dfa3605 : Reparent Nonlinear beamformer under beamforming interface.
    • bf395c1 : Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
    • caae5d4 : Bye request should use POST not GET
    • 190c3ca : Register sample rate of Audio RED in RTPPayloadRegistry.
    • 79064e5 : Fix crash on decode found by fuzz tester.
    • 3fbf99c : Refactor common_audio/vad: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT
    • 855acf7 : Remove video from WebRTC Android example. This is in preparation to remove the use of the old Video Api and the use of the old video capture module on Android in particular.
    • d4362cd : Reject StreamParams with RTX SSRCs not in ssrcs.
    • a49f515 : Roll chromium_revision da9a1c0..4d63ee8 (321718:322012)
    • 1ccd8b4 : Refactor common_audio/signal_processing: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT
    • 245989b : Address comments from cr 43769004. - Remove unnecessary hop to worker from OnChannelRequestSignaling_s. - Remove now-not-needed component param. - Update documentation.
    • 0e209b0 : Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/.
    • e61c64d : Delete NullVideoRenderer
    • 07a4ba5 : Simulcast settings for 1080p. Using same bit rates for all 3 modes since only one is used in reality, and the plan is to unify them.
    • ac27e20 : Delete VideoAdapter::AdaptFrame
    • 45636ec : Post Git switch: Update codereview.settings and remove drover.properties
    • 68a5418 : Enable PENDING_REF_PREFIX in codereview.settings.
    • 4d14592 : rtc::Buffer: Restore length method for backwards compatibility
    • deafa7b : Remove I420VideoFrame::SwapFrame
    • 2d2a30c : Remove I420VideoFrame::CloneFrame
    • 0b52ceb : Improve logging and add DCHECKs in codec database.
    • eebcab5 : rtc::Buffer: Rename length to size, for conformance with the STL
    • e815290 : Update README instructions for Android AppRTCDemo.
    • a5f6fb5 : Permit single-stream max bitrates above 2000k.
    • a197a5e : Update libsrtp includes in preparation of roll into Chromium.
    • a3ffc56 : Allow setting thread priorities in Chromium on all but linux platforms. The previous check was overly broad, so narrowing it down to linux only.
    • 39fc1d3 : Disable PeerConnectionClientTest.testLoopbackVp9
    • 0b44b58 : Limit disabling of PeerConnectionEndToEndTest.Call to Windows
    • 64eb2ff : iOS library build script
    • 9509fbf : Split EventWrapper in twain. I'm splitting the timer functions in EventWrapper into a separate interface. - Users of the timer functions have different needs than users of a generic event - Providing a default implementation for EventWrapper that simply uses rtc::Event.
    • 82e8ae4 : Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest
    • 2b4ce3a : Convert webrtc/video/ abort/assert to CHECK/DCHECK.
    • 41d2bef : Limit RED audio payload to narrow band.
    • 1596a4f : Temporarily disable SetPriority when building with Chromium. This is due to errors we were hitting with Chromium's sandbox policy for pthread_setschedparam.
    • d4e7d49 : Scaler: Recycle allocations using buffer pool.
    • 09b6ff9 : Disable PLC for iSAC
    • ee0c5af : Remove unused version.py script.
    • aa0bbab : Fix build failure
    • a4bef3e : AcmReceiver: use std::map instead of an array to keep the list of decoders
    • 3335a4f : Prevent asserting on unset start bitrate.
    • 50ed0d9 : Roll chromium_revision 6311617..da9a1c0 (321517:321718)
    • e5e92bd : Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix)
    • cfde27e : Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows.
    • 38492c5 : Re-land 8810 "- Add a SetPriority method to ThreadWr..."
    • 90a1cb4 : Revert 8810 "- Add a SetPriority method to ThreadWrapper" Seeing if this is causing roll issues.
    • b789f62 : Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..."
    • 0c34001 : Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine." content_browsertests started failing around the time the change landed and rolls are failing now. I'm going to try rolling this back, start a roll, and then re-land.
    • 346a64b : Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default. So in the HandleStreamFormatChange() callback, we need to re-initiate the playout as same as what we do in InitPlayout(). Here we merely copy those codes out from InitPlayout() into a new SetDesiredPlayoutFormat() function for the invoking from the two places. Previously, HandleStreamFormatChange only re-creates the AudioConverter, which is not enough. We also need to reset the buffer size and refresh the latency.
    • 4553941 : Document the 'int' return value of Resampler methods.
    • 3200a64 : Minor fix for MIPS Android build.
    • 4ddc938 : Support VP8 hardware encoding and decoding on IA devices.
    • b9557a9 : Fix code to handle crashes for non-VP8.
    • b6817d7 : - Add a SetPriority method to ThreadWrapper - Remove 'priority' from CreateThread and related member variables from implementations - Make supplying a name for threads, non-optional
    • 66df3cf : Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
    • 8296ec5 : Fix heap-use-after-free in WebRtcVideoEngine2.
    • a3209a2 : Release buffer pool in Vp8DecoderImpl::Release().
    • 8904290 : Make screenshare target bitrate experiment always on
    • d9c5024 : Roll chromium_revision bd49b12..6311617 (320783:321517)
    • 9f9ea7e : Clean up webrtc external capture. This cl removes the dependency to the external capture module if external capturing is used in webrtc. It also removes two external capture methods that is not needed. Further more it adds I420VideoFrame::Create that takes a pointer to packed memory as input.
    • 443ad40 : Remove FullStackTest frame pointer handles.
    • 6231fb6 : Prevent crashes when copying a zero-size frame.
    • 6069032 : Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32
    • 4ab23d0 : Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32
    • bd8c865 : Remove build-time beamformer flags.
    • 04c5098 : Add the Ooura FFT to RealFourier.
    • ba86031 : Whitespace change to trigger new Git pollers (2).
    • cf3fb9b : Whitespace change to trigger new Git pollers.
    • 80d9aee : Adds full-duplex unit test to AudioDeviceTest on Android
    • 361981f : Use scoped_ptr for ThreadWrapper::CreateThread.
    • c7d5a73 : Disable flaky test on DrMemory bots
    • 27c0be9 : Remove ThreadObj #define and kThreadMaxNameLength from thread_wrapper.
    • 0c26299 : Disabling two flaky tests in libjingle_media_unittest.
    • 17c64d1 : Revert "Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame"
    • c7157da : Use atomic operations for setting/reading the trace filter. The filter is currently being set and read by a number of threads and tripping up tsan.
    • 9afaee7 : Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal()
    • d21406d : Remove command-line tool 'video_coding_test'.
    • c4709a2 : Split C++ class from macro overrides to fix Chromium build
    • 5506a93 : Expose ViECaptureImpl::DisconnectCaptureDevice() to JNI of WebRTCDemo and call it before releasing camera to deregister the corresponding framecallback. Also stop camera after stop remote rendering as the correct termination order.
    • 8cc47e9 : Objective-C readability review.
    • 2a8a46d : vp8: Add missing call to SetUsageMessage().
    • 8f76cd2 : Renaming neteq_opus_fec_quality_test.
    • 840da7b : Implement Rotation in Android Renderer.
    • 143451d : Base start bitrate on last observed bitrate.
    • 5a477a0 : DCHECK frame parameters instead of return codes.
    • 4346d92 : Use SendTimeHistory to keep track of send times in simulations.
    • f189933 : Removing henrik.lundin from OWNERS in video_coding/*
    • af612d5 : Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
    • 6dba1eb : Make AudioDecoder stateless
    • 14ee8cc : WebRtcVideoFrame: Support odd resolutions
    • fc562e0 : Delete ACMGenericCodec::Encode and use AudioEncoder::Encode directly
    • 019955d : Revert 8749 "We changed Encode() and EncodeInternal() return typ..."
    • 3fffd66 : Revert "Implement Rotation in Android Renderer."
    • 835ec63 : Implement Rotation in Android Renderer.
    • 52cd828 : Allow webrtc external encoder factories to declare encoders have internal camera sources.
    • edd517b : Fix FYI build - add a missing include to event_tracer.h in system_wrappers.
    • 54d072e : Add CVO support to video_coding layer.
    • 63a1097 : Remove troublesome Windows line ending.
    • 462dbcf : Fix bug in Transport where channel_.clear() was being called without a lock. Looks like this snuck in between misaligned braces.
    • b493cb4 : Add storage alignment fix for opengles2.0 for iOS
    • da4fcc4 : Add minor fixes to video_capture_ios.mm in order to make it more robust.
    • 2161234 : Add new features to AppRTCDemo from private repo.
    • 779c3d1 : Use ByteReader/ByteWriter instead of rtputility and manual shift/add.
    • 09098da : Fix screenshare loopback target bitrate which isn't correctly configured
    • 25819b8 : Revert 8753 "Use atomic operations for setting/reading the trace..." Caused VP9 test to fail on TSAN and doesn't build in some configuration due to "../webrtc/base/criticalsection.h:181:12: error: cannot compile this atomic library call yet" :-(
    • b91d0f5 : 1. Have IPIsPrivate calling IPIsLinkLocal 2. Also check the Mac based IPv6 3. move the ip filtering into createnetwork. It shouldn't be done in IsIgnoredNetwork as the IP inside that could change later.
    • 3093390 : Parsing of transport wide sequence number rtp extension header. Plus some refactoring to correctly handle padding.
    • 1e69252 : Write commit position as a comment in Chromium DEPS.
    • 7c64ed2 : Move trace_event and associated files to webrtc/base. Also starting to use TRACE_EVENT from thread.cc in webrtc/base, to track Invoke() calls.
    • 7c112f3 : Adding build_opus as a switch in GYP.
    • c383c24 : Use atomic operations for setting/reading the trace filter. The filter is currently being set and read by a number of threads and tripping up tsan.
    • a846371 : Modify EventPosix to prevent spurious wakeups.
    • a78a94e : Fix RateTracker to set an initial reference time when first updated.
    • e155dbe : VP8/9EncoderImpl::Encode: Check resolution of input I420VideoFrame
    • 0cb612b : We changed Encode() and EncodeInternal() return type from bool to void in this issue: https://webrtc-codereview.appspot.com/38279004/ Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info.
    • 73d763e : Add I420 buffer pool to avoid unnecessary allocations
    • ae222b5 : Remove dead code in WebRtcVideoEngine2 unittests.
    • 858024f : WebRtcVideoFrame: Initialize members in empty constructor
    • 646eeac : Roll chromium_revision 8d51d96..bd49b12 (320682:320783)
    • 06d9390 : Adjust a threshold in VP9 test.
    • 592470b : Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession.
    • 12e7951 : Remove libvpx suppression due to fixed bug.
    • 6ad507a : Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE.
    • 4eeef58 : Remove a hacky dependency of BaseChannel on BaseSession by moving the handling of DTLS setup failure into a signal on BaseChannel rather than a method call on BaseSession.
    • c04a97f : Move from BaseSession::GetStats to WebRtcSession::GetTransportStats
    • aba9219 : Change ThreadPosix to use an auto-reset event instead of manual reset now that we know the problem we had with EventWrapper::Wait was simply a bug in the EventWrapper. Also removing |started_| since we can just check the thread_ instead.
    • 02d166b : Fixing a race condition in ACMGenericCodec
    • 3f11823 : Disables SW AEC when built-in AEC is enabled
    • 8bd2f40 : Remove code related to REMB suppressor experiment.
    • 2056ee3 : Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
    • 93d9d65 : I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments.
    • 2dc5fa6 : Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*.
    • 7f7d7e3 : Prevent crash in NetEQ when decoder overflow.
    • 4b89aa0 : Change StatsCollector to use DCHECK instead of ASSERT.
    • eed2fca : Roll chromium_revision 00e438c..8d51d96 (320241:320682)
    • 2d25b44 : Check associated payload type when negotiate RTX codecs.
    • eb44fd6 : Add flag to always close previous roll + minor refactor
    • c29f7f3 : Disable assert for nr of threads in PeerConnectionTest.java. This test is flaky so we need to figure out a better way to do it. I've documented what we've observed and added a todo for myself to figure out a solution.
    • 6107ba1 : Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame
    • f1f558c : Fix AppRTCDemo and AppRTCDemoTest builds.
    • d83f4ef : Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
    • b01c707 : Use a NULL session in unit tests that don't actually use the session.
    • b4aac13 : Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well.
    • 990a00c : Remove unused transport code.
    • c449c20 : Flag to wait for trybots to complete.
    • bc2bb34 : Refactor audio_coding/codecs/isac: Removed usage of macro WEBRTC_SPL_MUL_16_16
    • 9b2e114 : Supporting Opus DTX in Voice Engine.
    • dd0292a : Send to CQ by default and add --no-commit flag + cleanup.
    • 503a9e8 : Make AppRTCDemoTest pass without Internet connection.
    • 0c5b137 : Remove support for iSAC RCU
    • 9f41810 : Roll chromium_revision 87ce36b..00e438c (319600:320241)
    • 8372888 : Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns."
    • 0f663de : Rename Beamformer to NonlinearBeamformer.
    • 8663973 : Remove thread id from ThreadWrapper::Start().
    • e534086 : Clean up LappedTransform and Blocker.
    • 3d3c005 : Fix Android peer connection client instrumentation tests.
    • fde1de9 : Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
    • 00c509a : Add concept of whether video renderer supports rotation.
    • 04cd698 : Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns."
    • fdd1057 : Add CVO support to Vie layer.
    • 4f85288 : Socket options are only applied when first setting TransportChannelImpl.
    • 93604da : Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
    • d390029 : Use a variant for storing stats values in StatsCollector code.
    • c339276 : Fixing r8698.
    • e16bfde : Adding flag to force Opus application and DTX while toggling.
    • 75b7f17 : Temporary interface change to StatsReport::Id.
    • afdd5dd : Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame""
    • b73758d : Clean up VideoRenderFrames
    • d71fc87 : Make auto-roll script work with separate Chromium checkouts.
    • cade82c : Refactor MediaOptimization protection methods.
    • 119c12f : Revert "Suppress memcheck error in video_engine_tests"
    • 5c9f69f : Update the dummy file_audio_video_device to allow empty file name
    • e9413c6 : Revert 8689 "Fix an issue in DtlsIdentityStore when the store is..."
    • 2a3942a : Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
    • d2c09dd : Make building openmax_dl conditional in gyp.
    • 8c5ea8a : Fix temporal layer log string.
    • ae1a078 : Convert AppRTCDemo and AppRTCDemoTest to proper GYP target.
    • f1182dd : Make sure input manager lock is accessed after channel manager lock.
    • b218ff5 : Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"
    • 370a72c : Remove frame copy from cricket::VideoFrame to I420VideoFrame
    • 5c72922 : Remove unused member functions in audio_device_mac.h, which would cause compiling warning with clang -Wthread-safety-anaysis. Reported and fixed by mozilla. Imported here(We don't have any problem since we suppressed those warning in r7961).
    • 4dccdff : Add unittest to check that ViECapturer does not hold on to frames after they have been delivered
    • e77c9c8 : Build WebRtcMediaEngine2 outside of Chromium.
    • 0d9bb8e : Remove the need for scoped_ptrI420VideoFrame in VieCapturer.
    • 9bfa5f0 : In r8605, DTLS is enabled by default for native webrtc. So we have to disable it explicitly in peerconnection example for loopback test.
    • ece4b28 : FecTest: Reduce loop over numMediaPackets in test_fec.
    • f4e2060 : Script to roll WebRTC in Chromium DEPS
    • fc51607 : Fix Android AppRTCDemo failure on devices with one or no camera.
    • 4052d88 : Remove GetLastRenderedFrame
    • 49d0d34 : Making sure neteq gets compiled with OPUS.
    • 51ccf37 : AudioEncoder: add method MaxEncodedBytes
    • d7452a0 : Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
    • 4c8b930 : Zero-initialize all members of EncodedFrame.
    • 74d4792 : Fixes issue in RunPlayoutWithFileAsSource related to uninitialized member
    • aa57702 : Removed texture_video_frame.h and webrtctexturevideoframe.h
    • 7ef8b12 : Refactor audio_processing/ns: Removes usage of macro WEBRTC_SPL_MUL_16_16_RSFT
    • b38b009 : Refactor audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16_RSFT
    • 1afbdc7 : Refactor audio_processing/agc: Removes usage of macro WEBRTC_SPL_MUL_16_16_RSFT
    • f9a75d9 : Revert "Add concept of whether video renderer supports rotation."
    • 60a2aa0 : Revert "Add concept of whether video renderer supports rotation."
    • 31d1646 : Add concept of whether video renderer supports rotation.
    • 0ad4893 : Add concept of whether video renderer supports rotation.
    • dad85aa : Chromium build fix: Include new .cc files in rtc_base
    • a3823e2 : Hide assembly symbols.
    • 67186fe : Fix clang style warnings in webrtc/base
    • 2989204 : Fix instability in peer connection client unit test.
    • 59140d6 : Remove VideoRotationMode to VideoRotation.
    • 600587d : Refactor audio_coding/neteq: Removed usage of macro WEBRTC_SPL_16_16_RSFT
    • c7faace : Roll chromium_revision e8ef1d1..87ce36b (319252:319600)
    • 474d1eb : Adds C++/JNI/Java unit test for audio device module on Android.
    • 1b32bbe : Removing private and unused method in RTPReceiver.
    • 6b56d07 : Revert 8632 "Enable isac NEON building on Aarch64"
    • 385b566 : Revert "Workaround Mac align bug for observer_ and crit_."
    • a50e6f0 : Move ownership of vie_encoders and vie_channels into the channel group.
    • a32f064 : Fix build configuration bug with debug builds.
    • 558dc40 : Reland 8631 "Speculative revert of 8631 "Remove lock from Bitrat..."
    • 679d2f1 : Disable CS_TRACK_OWNER on Mac in debug mode. Local testing indicates that the pthread_t member variable might be causing alignment problems on the Chromium bots. After landing this (and once the Chromium tree is open again), I'll try a roll again to see if this has an effect.
    • f696e49 : Re-landing perf improvement for libjingle logging after reverting the general change. This contains only a part of r8635 that I just reverted to unblock the roll.
    • 52130b6 : Revert 8635 "Make LS_ logging constants to match Chromium's logg..." LibjingleLoggingTests in Chromium started failing so more thought needs to be applied here. Would be good to get he perf improvement in though.
    • 92696cd : Speculative revert of 8631 "Remove lock from Bitrate() and FrameRate() in Video..."
    • dc08a23 : Fix H.264 start code position search.
    • 1af1391 : Remove WebRtcTextureVideoFrame
    • c2008a0 : RTCOpenGLVideoRenderer: Add support for padded frames
    • b4cd093 : Change the unintentioal CHECK to DCHECK in DtlsIdentityStore.
    • 66f153f : Make LS_ logging constants to match Chromium's logging constants when building with Chrome. This was causing logging to be done at incorrect levels and filters not work as expected.
    • a2a6fe6 : Reconfigure default streams on AddRecvStream.
    • bcead30 : Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
    • 75e850e : Enable isac NEON building on Aarch64
    • 0d5ea21 : Remove lock from Bitrate() and FrameRate() in VideoSender. These methods are called on the VideoSender's construction thread, which is the same thread as modifies the value of _encoder. It's therefore safe to not require a lock to access _encoder on this thread.
    • f98030b : Add intermediate TextureVideoFrame typedef for Chromium
    • 45cdcce : Remove TextureVideoFrame
    • 7158ec1 : Remove android-webrtc.mk
    • e41ec81 : Remove libjingle_root GYP variable
    • e9217b4 : Remove WebRtcACMEncodingType
    • 84f5309 : Roll chromium_revision e144d30..e8ef1d1 (318658:319252)
    • a743f6f : Widening memcheck suppressions for libjingle_peerconnection_unittest
    • 818c498 : Modify the simulcast encoder factory adapter to allow external encoder factories that support more than one codec.
    • 16a87b9 : Add VP9 denoiser test to videoprocessor_integrationtest.
    • 1d88394 : Add support for arbitrary array geometries in Beamformer
    • 0933d01 : Enabling common_audio building with NEON on ARM64
    • d7a212e : audio_processing/aec: Increased delay metrics aggregation window to five seconds
    • c3f15c0 : Fix scoped_ptrs in bwe_simulations.
    • 7430433 : Print better information during Chromium sync.
    • 2386d6d : Revert 8599 "Revert 8580 "Unify underlying frame buffer in I420VideoFrame and...""
    • 67a9e40 : Prevent encoding frames with wrong resolution.
    • 0305448 : Adding basic support for posting tasks to a process thread.
    • 658d201 : Allow VideoSender to be constructed on one thread but initialized and used for doing registrations, on another.
    • 7008f22 : Revert Clang roll in r8596 + add memcheck suppression.
    • 5af41aa : Fix uninitialized variable. If FindConstraint() returns false, we check |value| in two places and at that point, it can hold an uninitialized value. Caught by Linux Memcheck builder.
    • fa67463 : skip isac_neon if neon is not supported
    • bbce5ef : Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
    • d43b2c0 : Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
    • 86c33e3 : Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
    • 4536289 : Add CVO support to RTP sender side.
    • 61e00b0 : Create a in-memory DTLS identity store that keeps a free identity generated in the background.
    • 6daacbc : Set cpu_speed parameter for low resolutions, for non-simulcast.
    • 7b93ea1 : Remove DCHECK from common_types.cc
    • 4a4e688 : Remove dependecy on win32.h in criticalsection.h. This was causing build errors that we haven't fully figured out yet but somehow this caused override files to include the files they're supposed to override, which in turn included webrtc build files that then conflict with Chromium's configuration.
    • f7abb12 : Fix OVERRIDE-override again after reverting video frame cl.
    • 1f94407 : Revert 8580 "Unify underlying frame buffer in I420VideoFrame and..."
    • c86bbba : Add speech flag to EncodedInfo
    • 92f4018 : Start using std::map for Values in the statscollector. This is in preparaton for more work which will cut down on the string copying work we do.
    • 14665ff : Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
    • 792f1a1 : Break out allocation from BitrateController into a BitrateAllocator.
    • 61c22ac : Eliminate AcmGenericCodec::Add10MsData
    • f82109c : Initialize memory in I420VideoFrame unittest
    • 487afc7 : Always define RTC_NOTREACHED, not just in non-chromium builds
    • 9cd7c26 : Rename NOTREACHED to RTC_NOTREACHED to avoid name conflict with Chromium
    • 6dab6d7 : Let Chromium declare the mips_dsp_rev build variable.
    • 1d25c87 : Reland r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
    • 058b1f1 : Remove GetReceiveBandwidthEstimatorStats.
    • 7572d85 : rtc_unittests on Android
    • c98f6f3 : PRESUBMIT: Exclude overrides paths from source above GYP check.
    • fc2f146 : Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
    • 7bea1ff : Expose negotiated ciphers through stats API.
    • be77872 : Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
    • bbbdeed : Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
    • 369f682 : Create a in-memory DTLS identity store that keeps a free identity generated in the background.
    • c8895aa : Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
    • 8ad9660 : Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
    • bcef431 : Revert r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
    • 1fc28f2 : Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac
    • df512cc : Create a in-memory DTLS identity store that keeps a free identity generated in the background.
    • 982cd2a : Filter receiver-side DataCountersUpdated on SSRC.
    • b144b4b : Fixed bug in SendTimeHistory, where deleting packets via the getter would not update the oldest suence number.
    • 0561716 : Adding Opus DTX support in ACM.
    • a1c9803 : Fix crash in setPictureSize on Galaxy Nexus. This cl tries to find the best supported pictureSize before setting it. BUG=4197 R=magjed@webrtc.org
    • be00e3c : Make sure VideoFrameFactory handles rotated frames when scaling.
    • 9e5f941 : Remove webrtc/system_wrappers/interface/scoped_ptr.h
    • 1f914ec : Remove suppression for WebRtcVideoFrameTest::TestInit The problem is fixed in https://webrtc-codereview.appspot.com/41029004/.
    • db93b68 : Removing NetEq's direct dependencies on Opus headers.
    • cb04aa4 : WebRtcVideoFrameTest: Initialize memory to fix DrMemory error
    • 909f494 : Roll chromium_revision 2c3ffb2..e144d30 (317530:318658)
    • 1d82813 : Reland "Fix CVO in androidvideocapturer".
    • c9ce07e : Add Config option to enable 48kHz support in AudioProcessing
    • 0482d01 : Implement TraceCallback in a nested class of WebRtcVideoEngine. This is to fix a race that occurs in unit tests when the tests inherit from the engine class that also implements the callback interface for tracing. If tracing happens while the most derived class is still being constructed, we're in trouble.
    • 97ed2a4 : I420VideoFrame: Remove function ResetSize
    • 43f4a47 : Add more Android peer connection client unit tests:
    • 976c0f3 : audio_processing/aec: NEON code should not be invoked if it is detectable, but is not NEON
    • 48ac226 : Add support for writing h264 decoder input to file and parsing interleaved length/packet RTP dumps.
    • 3fe17d1 : Adjust a few thresholds for VP9 tests.
    • fd33293 : I420VideoFrame: Remove functions set_width and set_height
    • f1f0d9a : Remove WebRtcVideoEngine::SetVoiceEngine.
    • e8f50df : Remove avi recorder and corresponding enable_video flags.
    • f56c162 : Remove AudioCodingModule::Process()
    • 25dd1db : Fixed bug in test frame generator, causing incorrect reuse of frame object, in turn causing performance regression.
    • 60f9d6f : Revert "Add default implementation to VideoSourceInterface." Chrome test mock has been updated so VideoSourceInterface can now be pure virtual again. This reverts commit ed8d52378c43a7a93e0d2ca586486ca06db9eabe.
    • afa6d16 : Add a ToString() method to StatsReport::Value. This is an interface change only at this point which will be followed up by a matching change in Chromium that removes the dependency on the 'value' member variable. Once that's been done, I'll add native support for non-string types in the Value class.
    • 50b2295 : cricket::VideoFrameFactory: Don't overwrite frames in use
    • 24485eb : Remove last pieces of libjingle_unittest
    • 5cd6828 : Remove stale isolate files.
    • f35e4bc : Introduce a send time history class, keeping track of packet send times.
    • 59ae5ff : Filter logic for ip leak misses ::ffff:0.0.0.0
    • 2f6ae0d : audio_coding/codec/ilbc: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT
    • e1b84a0 : Fix a race reported by tsan. TSAN complains about this variable not having synchronized access, so I'm using atomic operations on it instead. There's no functional difference really though.
    • d68fa65 : Improve cleaning for Android demo applications
    • f7bb6e7 : Use new API from BoringSSL to get RFC name of cipher.
    • d312505 : Test to try to track down the alignment problem on Mac 10.9. There's no code change here, I'm rearranging member variables of the trace class and adding a sizeof check to the CriticalSection class + alignment attribute for the mutex, on Mac only.
    • 73acc15 : Revert 8538 "Reland "Fix CVO in androidvideocapturer."""
    • 3a93e33 : Reland "Fix CVO in androidvideocapturer."" This reverts commit b8bcf8cbbf84971e2ae26d91659afdc58617b054. after I fixed a rebase mistake. The fix is the delta between patchset 1 and 2.
    • b8bcf8c : Revert "Fix CVO in androidvideocapturer."
    • 02ed57b : Fix CVO in androidvideocapturer.
    • 41d8fda : VideoCapturerAndroid allocates direct buffers so that the frame buffers can be used in C++ without a copy. However byte[] array = ByteBuffer.array() seems to point to the beginning of the underlaying buffer and that is what the camera fills. But it turns out that ByteBuffer.arrayOffset() returns an offset and it seems like the pointer returned by jni-GetDirectBufferAddress(j_frame). This cl reverts back to pass the byte[] to c++ and use jni-GetByteArrayElements to get the address of the buffer.
    • 07dcf60 : Revert 8532 "Ensure only temporary IPv6 address is selected as t..."
    • 21ad375 : Ensure we set the right attrib for correct shader
    • 385a7ce : Ensure only temporary IPv6 address is selected as the best IP.
    • fbef5c6 : Remove lock in ViEFrameProviderBase::IsFrameCallbackRegistered.
    • 7400e0b : Revert "I420VideoFrame: Remove functions set_width, set_height, and ResetSize"
    • 4b3618c : Remove TraceImpl logging thread.
    • 6c2e506 : Workaround Mac align bug for observer_ and crit_.
    • 3985f01 : ProcessThread improvements.
    • f296859 : PeerConnectionClient.createPeerConnectionClient was calling new PeerConnectionParameters and PeerConnectionClient.createPeerConnectionFactory, .createPeerConnection with invalid arguments.
    • c68e0c9 : Fix cpplint warning in the previous cl to peerconnection client example.
    • abbdd52 : AudioEncoder: documentation fix
    • ea89495 : Remove {Is,Set}BlackOutput from VideoAdapter.
    • 3aca0b0 : Add 48kHz support to Beamformer
    • 9650ab4 : Fix case sensitivity of AppRTCDemo include dirs
    • 2a72c65 : Keep feedback params in SetDefaultEncoderConfig.
    • b1f0de3 : AudioEncoder: change Encode and EncodeInternal return type to void
    • 00b8f6b : Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
    • ac2d27d : Fix style violations in common_types.h and config.h
    • 891d483 : Wire up target_media_bitrate in VideoSendStream.
    • 9dd0ebc : Remove the default RTP module.
    • 3e6e271 : Implement CpuOveruseMetrics as callbacks.
    • 60f295f : Remove lsan suppression.txt
    • e723728 : Add p2p.gyp to rtc_base presubmit check exclusion.
    • 9a4410e : Implement adaptation stats in WebRtcVideoEngine2.
    • 38d9cc5 : Add back return statement after FATAL()
    • b5e60b6 : Remove non necessary check from WebSocket send function.
    • f09e7b8 : WebRtcVideoFrame: DCHECK exclusive ownership for non-const pixel access
    • 6c66163 : Fix TestScaler PSNR tests
    • 96abda0 : Removing FEC functionality from the default RTP module.
    • 9b969e1 : AudioEncoderCopyRed: CHECK that encode call doesn't fail
    • 749c602 : Moved gypi to avoid presubmit warning about '..' when touching the files.
    • 5c928eb : Let first packet through to avoid getting key frame requests (and no nacks) for EndToEndTest.ReceivedFecPacketsNotNacked.
    • 09c77b9 : Add decoder-timing stats to VideoReceiveStream.
    • c5558b7 : Remove AudioCodingModule's dependency on the Module interface
    • af82f75 : Let Add10MsData method do the encoding work as well
    • 4aef5fe : Add thread checks to the CaptureManager.
    • 8d350d4 : Add new AcmGenericCodecTest and verify output from Encode function
    • 1eda4e3 : Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"
    • 1e64263 : Thread-safe ChannelManager.GetSupportedFormats, used by VideoSource
    • 0a3ff79 : New AudioTrack implementation now works on pre-Lollipop devices.
    • 112f127 : Refactor how VideoCapturerAndroid delivers frames and is stopped. With this cl, video buffers are now allocated using direct buffers. These buffers are guaranteed to live as long as the capturer is running. We can now post frames in c++ from the Java thread to the c++ worker thread and let c++ post the buffers back when it has finished processing them.
    • d4dfba8 : iSAC Decode: Prevent Memcheck from complaining about uninitialized value
    • 87a592d : Fix dependencies of media_file module and move gypi into the right dir to avoid submit warnings referencing files with '..'.
    • a4623d2 : Fix H.264 HW decoding for Qualcomm KK devices.
    • 49096de : DCHECK send DataCountersUpdated for valid SSRCs.
    • 903182b : Revert r8476 "Set decoder output frequency in AudioDecoder::Decode call"
    • 3480728 : Swap decl-terms from juberti@ review.
    • 3630085 : Tested equiv classes of DTLS/SCTP.
    • 91d5230 : Renamed string and test.
    • c7848b7 : Added a separate DTLS/SCTP test.
    • a747093 : After another round of reviews.
    • 9616196 : Merging definitions of IsSctp.
    • 12aa8a6 : Post-rebase.
    • 1730869 : Added raw SCTP to IsSctp.
    • 871b1c3 : Review comments -- added IsSctp()
    • d7b6165 : Made DTLS/SCTP equivalent to UDP/DTLS/SCTP when comparing session descs in tests.
    • ec97c65 : Attempt on read-only acceptance of -12.
    • b9c18d5 : Set decoder output frequency in AudioDecoder::Decode call
    • f88791d : AudioEncoderCng: CHECK that encode calls don't fail
    • 5e3fea1 : Fixing WebRTC engine demo JNI symbol export.
    • a30f007 : Fixing incorrect memset in mock class.
    • a5de951 : Make Options public and not package access in pc factory.
    • db8e605 : Break out BWE test models to separate files
    • ccd7c7c : Remove more unused code in ACM
    • 13ca5f6 : AudioEncoderOpus: CHECK that encode call doesn't fail
    • d324546 : Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : * Move constants into the files/functions that use them * Declare variables in the narrowest scope possible * Use correct (expected, actual) order for gtest macros * Remove unused functions * Untabify * 80-column limit * Avoid C-style casts * Prefer true typed constants to "enum hack" constants * Print size_t using the right format macro * Shorten and simplify code * Other random cleanup bits and style fixes
    • 7227391 : Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
    • b28474c : Add H.264 HW encoder and decoder support for Android.
    • 77e11bb : Wire up preferred/nominal_bitrate to stats.
    • f3a306b : g722: Enhanced documentation. Added CHECK.
    • 2acec4c : Enhanced documentation. Replaced DCHECK with CHECK.
    • 962c624 : Refactoring WebRTC Java/JNI audio track in C++ and Java.
    • 2ad3bb1 : Reland patch for Switch default color format to YV12 on Android. The new since the previous patch is that we ignore all resolutions with width % 16 != 0 since they are not tightly packed.
    • 8278c07 : Enable NACK under SendsAndReceivesH264.
    • fa58745 : Delete all codec-specific subclasses of ACMGenericCodec
    • 2a5cfc2 : Replaced unnecessary check with an explicit CHECK. WebRtcIlbcfix_Encode method that is called returns an error code only if a packet with more than 3 frames is passed, which is illegal.
    • 343096a : Fix incorrect rtx config in full_stack tests.
    • 1467421 : Fix for flaky test: VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent.
    • 50e2816 : Move SetTargetSendBitrates logic from default module to payload router.
    • a43fce6 : Add functions rtc::AtomicOps::Load and rtc::RefCountedObject::HasOneRef
    • 2af3057 : Revert "When clearing the priority message queue, don't copy an item to itself."
    • 2bffc3c : When clearing the priority message queue, don't copy an item to itself.
    • d3a487c : Exclude end-to-end test RestartingSendStreamPreservesRtpStatesWithRt on memcheck.
    • 3c4668e : Amend CpuMonitor fix.
    • f906e55 : Add CpuMonitor to Android ApprtcDemo
    • 7ac374a : Fix shutdown race for ViEEncoder when there is a frame in the encoder.
    • dc77d74 : Disable FullStackTest.ForemanCifPlr5 temporarily while investigating flakiness.
    • ec45e3b : Fix test race in GetStatsMultipleSendStreams.
    • 804eb46 : Change default from GICE to ICE5245 for SDP offers
    • d3d3baa : Copy SetThreadName from webrtc/base/thread.cc into thread_win.cc (webrtc/system_wrappers/source/thread_win.cc). It would be good to consolidate these helpers at some point.
    • 661af50 : Small Beamformer optimization
    • cce874b : Fix libjingle_media_unittest codec comparison issue
    • bc6961f : Make webrtc 50 KB smaller by not inlining Codec.
    • e07710c : Make SendCodec() lock-free.
    • be29b3b : I420VideoFrame: Remove functions set_width, set_height, and ResetSize
    • be96bfb : Re-land "Switch to using AudioEncoderIsac instead of ACMISAC"
    • 1ed6224 : Revert r8430 "Remove dead stats from Video{Sender,Receiver}Info."
    • 2877552 : Fix a problem with reading uninitialized memory in ACM
    • 8ad05b7 : Remove dead stats from Video{Sender,Receiver}Info.
    • 1d0fa5d : Add RtcpPacketTypeCounter stats to new API.
    • 5060412 : Method WebRtc_g722_encode that is eventually called always returns non-negative integer (internal counter)
    • 47d657b : Remove Set/Get sending status from the default RTP module.
    • 32c784c : ViEExternalRendererImpl: Remove dependency to webrtc::VideoFrame
    • 3db042e : Stop AndroidVideoCapturer asynchronously. The purpose is to avoid a deadlock between the C++ thread calling Stop and the Java thread that provides video frames.
    • 2548406 : Add empty files to implement a in-memory DTLS identity store without breaking Chromium build.
    • 652bc37 : Adding two new stats to StatsReport.
    • a744a28 : Templatize and clean up codec wildcards.
    • 30540fe : Initialize RTPVideoHeader fields to correctly set simulcastIdx for non VP8 codecs.
    • 9dfe7aa : Fix WebRTC IP leaks.
    • 931e0cf : Fix WebRTC IP leaks.
    • f358aea : Fix WebRTC IP leaks.
    • 18c9247 : Move Android MediaCodec encoder and decoder factories to separate files.
    • 88828e7 : Fix I420VideoFrame unittests
    • c0bd7be : Adding two new stats to VoiceReceiverInfo
    • 8fbdcfd : Revert "Switch default color format to YV12."
    • b255865 : The PCM codecs can never fail, so we don't need to check the return value
    • 78619e2 : Revert of r8378 "Switch to using AudioEncoderIsac instead of ACMISAC"
    • 1c3e728 : Switch default color format to YV12. Currently N21 is used per default. But according to http://developer.android.com/reference/android/graphics/ImageFormat.html#YV12 YV12 has been mandatory to support since api level 12. Since YV12 and I420 is the same except for the order of planes, this format is cheaper to use.
    • 635838b : Re-implementing AcmOpusTest as AcmGenericCodecOpusTest
    • f68e186 : Remove EnableMirroring and MirrorRenderStream
    • 131bea8 : Offline screenshare quality test, plus loopback.
    • 0521127 : AudioEncoder: Rename virtual accessors to CamelCase
    • cc483b7 : Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)
    • b4987bf : Send black frame with previous size when muting.
    • 7d721ee : Adding speech_expand_rate to NetEQ Network Statistics.
    • 3864363 : cricket::VideoFrame: Refactor CopyToBuffer into base class
    • dd4a8da : Remove DISABLE_YUV flag
    • 97aaf68 : Bump to version 42.
    • bfa3c72 : Don't call g_thread_init on glib =2.31.0
    • e9facf8 : Add range checks in a variety of places where the values will subsequently be expected to be 0-127.
    • 27669f3 : Apply good settings to Beamformer
    • b08f404 : Fix issue 4061.
    • 0abc601 : Remove SetCaptureDelay from the RTP module.
    • 7663684 : Implement the Nada rmcat proposal within the simulation framework.
    • 71b35a4 : iLBC: Use uint8_t[] for byte arrays
    • 640313c : WebRtcVideoCapturer: Remove dead code |OnIncomingCapturedEncodedFrame|
    • 7a91acb : ViECapturer: Remove unimplemented function declaration |DeliverCodedFrame|
    • 1a38a51 : Add default implementation to VideoSourceInterface of Stop and Restart. This is to make sure Chrome does not break when rolling. This should be reverted once Chrome has been updated.
    • a28a91d : Fix data race for RTCPReceiver stats callback.
    • 8f605e8 : Add VideoSource::Stop and Restart methods. The purpose is to make sure that start and stop is called on the correct thread on Android. It also cleans up the Java VideoSource implementation.
    • 959dac7 : VideoCaptureImpl: Remove unused member variable |_capture_encoded_frame|
    • 4dd40d6 : Signal threads for faster receiver destruction.
    • 0a7d4ee : Remove usage of BitrateController in VoiceEngine.
    • f9b5c1b : Removing CELT.
    • 2c1bcf2 : Adding decoded_fec_rate to NetEQ Network Statistics.
    • 290cb56 : Remove TimeToSendPacket and TimeToSendPadding from the default module.
    • c0fc4dd : Add 'mac_x64' trybot to default set.
    • 86196c4 : Setup encoders inexpensively before first frame.
    • 34509d9 : Fix an issue with comfort noise in ACMGenericCodecWrapper
    • e9f0f59 : Enable bitrate probing by default in PacedSender.
    • fbc347f : Re-land r8342 "Switch to using AudioEncoderIsac instead of ACMISAC""
    • ce22f13 : GN: Changes for vp9, opus and direct trace
    • e35fa96 : Move isacfix.gypi and isac.gypi
    • 0200f70 : Set webrtc_rtp category to be disabled by default.
    • 14b0279 : Break out code from bloated files in the BWE simulator.
    • 0f7f161 : Add audio_coding module OWNERS file.
    • 4dc0003 : Revert r8342 "Switch to using AudioEncoderIsac instead of ACMISAC"
    • 30142bb : Add arraysize to overrides to avoid macros redefinitions in Chromium
    • d3b453b : Remove the incremental IP address behavior from virtualsocketserver
    • 3341b40 : Fix bug parsing media descriptions: the final field isn't a codec type for any of DTLS/SCTP, SCTP, or SCTP/DTLS.
    • 92a19bc : Simplify mask calculation
    • 56cb0ea : Add support for bi-directional simulations by having both an uplink and a downlink.
    • d5ce2e6 : Remove EventWrapper::Reset().
    • 5a7dc39 : This is a code clean up. No functional change intended.
    • a8cc344 : Allowing RED decoding for Opus.
    • 96e4db9 : Split peerconnection_jni.cc into separate files. For now: java_helpers - JNI convenience functions etc. Can in theory be moved to libjingle / webrtc general one day. classreferenceholder - app/webrtc specific Java class loader. androidvideocapturer_jni - the jni part of the video capturer I added. peerconnection_jni - all the rest.
    • 8db5854 : Fix potential flakiness in voe_auto_test.
    • 2adf4c4 : Re-enable BWE tests using baseline files.
    • 58f6f01 : WebRTC now compiles for enable_android_opensl=1.
    • 40fdb8a : Remove flaky test cases from peerconnection_unittest. The chain of API calls is tested from top to bottom anyway.
    • ba97ea6 : audio_coding/codec/ilbc: Removed usage of macro WEBRTC_SPL_MUL_16_16
    • 2bd299a : Remove call to RtpRtcp::RegisterSendPayload for the default RTP module.
    • 40367f9 : Remove default video encoders for new video API.
    • 94eb9a6 : Whitespace change to test gsubtreed.
    • e388c19 : Fix start bitrate settings for VP9 codec in AppRTCDemo.
    • bb1219e : Add a unit test for callbacks with empty frames and fix bug in code
    • e012643 : Remove temporary GYP targets
    • aafbec1 : Remove ViENetwork::SetBandwidthEstimationConfig() interface since dynamically changing BWE settings isn't necessary now that AIMD is the default.
    • 503c336 : Re-enabling LocalP2PTestAnswerVideo and LocalP2PTestAnswerAudio test cases in peerconnection_unittest.
    • a9eaeeb : Fix problem where Android VoE can not record on multiple channels.
    • 7c4d20f : Remove potential deadlock in RTPSenderAudio.
    • ff689be : Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
    • 9e4e524 : Use an external-only VideoRenderModule in Call.
    • a4ef2ce : Remove getting max payload length from default module.
    • 006521d : Makes libjingle_peerconnection_android_unittest run on networkless devices.
    • 3ee4fe5 : Re-land: Add API to get negotiated SSL ciphers
    • 76b4ac9 : Switch to using AudioEncoderIsac instead of ACMISAC
    • 6c68c85 : Switch to using AudioEncoderOpus instead of ACMOpus
    • 1226e92 : CVO capturer feature: allow unrotated frame flows through the capture pipeline.
    • dc7b022 : CVO capturer feature: allow unrotated frame flows through the capture pipeline.
    • 20e8f22 : CVO capturer feature: allow unrotated frame flows through the capture pipeline.
    • 073dd7b : WebRtc_GetCPUFeaturesARM is only available on android
    • a98e796 : Remove default RTP module functionality for setting CSRC.
    • a6e8ceb : Fix false positive DHECK in event_posix.cc
    • 11426dc : Don't rely on webrtc/base/scoped_ptr.h to include stuff for you
    • fbcb5ce : Remove VideoSendStreamTest.ProducesStats.
    • 9d94a0c : Switch to QueueUserAPC for shutting down the thread (no event needed). Also actually specifying the reserve stack size.
    • fddeaf5 : Switch to using AudioEncoderG722 instead of ACMG722
    • 83bc721 : Add Android specific VideoCapturer. The Java implementation of VideoCapturer is losely based on the the work in webrtc/modules/videocapturer.
    • c18957e : Make Git ignore in resources more fine-grained
    • 354becf : Remove Git ignore exclusion of .sha1 files
    • 7cc92aa : Use WebRtcVideoRenderFrame for texture frames.
    • 62f6e75 : Refactoring WebRTC Java/JNI audio recording in C++ and Java.
    • c2d0473 : Switch to using AudioEncoderPcm16B instead of ACMPCM16B
    • f58fe0a : Rename GYP and GN targets for video capture+render.
    • 2c29c2e : C++ readability review for ajm.
    • 5d60895 : Fix bug when there are no blocks in a chunk in Beamformer
    • bc35703 : Add a method to remove an existing renderer from the internal list of Android renderers.
    • d35a5c3 : Make ChannelBuffer aware of frequency bands
    • d7472b5 : base/arraysize.h: We use size_t, so need to include stddef.h
    • 91ba79a : Make sure that the norms are positive in Beamformer
    • b6856d2 : Apply mask smoothing in Beamformer
    • 8da96ac : Switch to using AudioEncoderIlbc instead of ACMILBC
    • 1a072f9 : Address comments from previous review round for rtc::Event.
    • f4c10d2 : Always use DeliverI420Frame in WebRtcVideoEngine.
    • 027e113 : Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig.
    • 30015e3 : Fix bug in EventPosix where we'd miss a set event. In cases of timeout or error, we could change the state of the event to 'down' (unset) and subsequently never satisfy a Wait() for a given Set().
    • 648f5d6 : pcm16b: Make input arrays const and use uint8_t[] for byte arrays
    • 948d617 : Create a separate thread for pacing.
    • c11348b : Fixing a bug in expand_rate calculation for stereo signal.
    • 8e612ab : Remove voice_engine_ member variable and GetVoiceEngine() from ViEChannelManager.
    • 5b8f3e0 : Roll chromium_revision 598c3e9..601e6f3
    • 44ae4c8 : Support using VP9 video codec in AppRTCDemo.
    • f7e6cfd : Add CHECK to EventWrapper to see if there's a subtle bug there or not.
    • 669bc7e : Modify default field trial implementation to allow WebRTC client to turn on feature code.
    • 11c5db0 : Revert 8273 "Temporarily change ThreadPosix to CHECK (crash) if ..."
    • 0d852d5 : Use VideoReceiveStream as an ExternalRenderer.
    • d6e25a5 : Revert r8297 "Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig."
    • 03c1c10 : Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig.
    • 53d9012 : Clean kForever from basictypes and move it to the interfaces that actually have it.
    • e01bae2 : Fixing a nit
    • 1c6239a : G711: Make input arrays const and use uint8_t[] for byte arrays
    • d0165c6 : Use a manual reset event in PosixThread. This fixes occasional hangs we've been seeing in the past few days. I'm using rtc::Event instead of the EventWrapper, so I'll wait with landing this cl until I've made that change in a separate cl.
    • 4c0fd96 : Move rtc::Event to rtc_base_approved. We need an event implementation in WebRTC that allows us to specify whether it's manually reset or automatically. EventWrapper currently doesn't support it and it adds a heap allocation + vtable, so rtc::Event is the lighter of the two.
    • 8cf9bdb : Remove USE_WEBRTC_DEV_BRANCH.
    • 2b69eab : Restructure GYP for vp9, opus and direct trace
    • f31f56d : Remove default arguments in EncodedImageCallback.
    • 6c930c7 : Cleanup: unify rotation to be enum based instead of int for degree.
    • 7a57f8f : Reland 8203 "Reducing locking in OveruseFrameDetect..." The issue that was causing the thread checker to report error, turned out to be unrelated.
    • 103f328 : Fix the binary layout of ProcessThreadImpl. We apparently hit an obscure problem on mac where seemingly an unaligned mutex causes memory corruption. The effect was that the |modules_| list became corrupt and we crashed. At this point I'm not exactly sure what the alignment requirements are but for now, I've fixed up the layout in a way that doesn't cause these same issues.
    • ec499be : Increase testclient timeout from 1 to 5 seconds
    • fe19699 : Revert 8260 "Base RWLockWrapper on rtc::SharedExclusiveLock." Unfortunately this caused channel teardown to hang. More details in email(s).
    • 2eb1660 : Switch ThreadCheckerImpl over to using PlatformThreadRef. Like PlatformThreadId, this type is borrowed from Chromium. The difference between the two is that PlatformThreadRef is pthread_t on posix platforms. On Windows PlatformThreadRef and PlatformThreadId are the same thing.
    • 2bf0e90 : Revert 8275 "This CL adds an API to the SSL stream adapters and ..."
    • 1d4830a : Disable ProcessThread tests that are dependent on timing. Some of the bots are too slow for the tests to make much sense as they are.
    • 95a32ec : Revert 8271 "VirtualSocketServer out-of-order issue with closing..."
    • 2a44be9 : Normalize delay-and-sum mask in Beamformer
    • 799e667 : Add high frequency correction to Beamformer
    • 0c7ec77 : Cleanup: unify rotation to be enum based instead of int for degree.
    • 110443a : Cleanup: unify rotation to be enum based instead of int for degree.
    • 1d11c82 : This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
    • 63da1dd : audio_processing: Now records mic volume level also when using new AGC
    • ccd7e99 : Temporarily change ThreadPosix to CHECK (crash) if we ever spend more than 30 seconds waiting for thread shutdown. There are cases on build bots where it looks like we're hitting this problem, but reproducing locally has been a struggle.
    • 13a0e18 : Temporarily disable a couple of ThreadChecker tests on Mac.
    • 4770437 : VirtualSocketServer out-of-order issue with closing TCP sockets
    • 9baa9ca : Add libjingle_peerconnection_so.so to Java test dependencies. This fix a problem where the Java test is not dependent on the so file.
    • b5a1252 : Hack to work around the current issues with rolling WebRTC into chromium. In order to figure out the issue with the Mac 10.9 debug bot, this patch disables the ThreadChecker class on Mac in debug builds. For diagnostic purposes, it instead prints out when there's a thread mismatch. I'm also adding a DCHECK in case fetching the current thread id ever returns 0.
    • 751a365 : Switch to using AudioEncoderPcmU/A instead of ACMPCMU/A
    • 02270cd : Implementing a packet router class, used to route RTP packets to the sending RTP module for the specified simulcast layer a frame belongs to. This CL also removes the corresponding functionality from the RTP RTCP module and fixes lint warnings in the files touched.
    • 10a9e92 : Fix delete of stack allocated object causing test crashes.
    • 4b320cf : Revert "Cleanup: unify rotation to be enum based instead of int for degree."
    • fb609a1 : Wire up new feedback format by introducing a FeedbackPacket type.
    • 353c8b8 : audio_processing/agc: Changed to correct include path in agc_unittests
    • bc3241a : Update ProcessCallAfterXms to better match the performance of our faster bots. Previously I had made sure these tests didn't flake out on our slow trybots, but apparently I need to do the same for the fast bots :)
    • 0c3e12b : Revamp the ProcessThreadImpl implementation.
    • 7502543 : Base RWLockWrapper on rtc::SharedExclusiveLock.
    • 5e05731 : Roll chromium_revision cd35af6..598c3e9
    • 57ac2c8 : Default destination used by c line should be IPv4 only to avoid parsing error in legacy client.
    • 3e733a4 : Cleanup: unify rotation to be enum based instead of int for degree.
    • 74d2788 : Remove defined(__cplusplus) tests in C++ code.
    • f45c8ca : Reland r8248 "Introduce ACMGenericCodecWrapper"
    • ec4521c : Clean up Beamformer initialization
    • e69220c : Fix the value of the first byte of nal unit generated by fake H.264 encoder.
    • f693229 : Fix Android video renderer to support video frames with stride width.
    • cc64a9c : voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric
    • 4b9622f : Roll gtest-parallel.
    • 3a87630 : Revert r8248 "Introduce ACMGenericCodecWrapper"
    • af8c13f : Introduce ACMGenericCodecWrapper
    • 5d32f43 : Disable CondVarTest.InitFunctionsWork. The order of Sleep/Wake calls doesn't seem to be guaranteed, so this test is flaky.
    • 877ac76 : Cleanup and prepare for bundling.
    • cf7efeb : Add new AudioEncoderOpusTest
    • 520a69e : Revert 8238 "Add RefCounting for TransportProxies"
    • 875c97e : Remove SetNotAlive method from the thread class. Also cleaning up methods with the same name in other classes that are derived from the above method.
    • c5f6971 : Revert 8237 "Cleanup and prepare for bundling."
    • dc096f2 : system_wrappers: Disabled flaky test CondVarTest.PassBatonMultipleTimes
    • 4414939 : Add method for incrementing RtpPacketCounter. Removes duplicate code.
    • e250667 : Add RefCounting for TransportProxies
    • af01d93 : Cleanup and prepare for bundling.
    • 322a564 : Fix datachannel stats id and timestamp.
    • d43bdf5 : Rewrite ThreadPosix. This is the same change as already made for Windows: https://webrtc-codereview.appspot.com/37069004/
    • bfdee69 : Roll chromium_revision 9070a80..cd35af6 (313233:314322)
    • 0ec50be : Changing include guard in frame_callback.h.
    • 200ac00 : Remove temp files in audio_processing_unittest.cc.
    • 0e8bf6c : Enable bitrate probing by default.
    • b1786db : audio_processing: Added a new AEC delay metric value that gives the amount of poor delays
    • 0e81fdf : Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
    • 19f3f71 : Fix apparent typo: int - char.
    • 946ad76 : Switched lists of packets to lists of packet pointers. Allows Packet polymorphism.
    • c957ffc : Fixed potential crash if rtp packet history is completely full.
    • c420a86 : Change name for local CriticalSectionScoped variable
    • a1dfbf1 : WebRtcG722_Decode: Input array should be const uint8_t[]
    • 026b892 : Using on an int8_t or uint8_t will output a character rather than a number. Places that do this need to cast to int to get the desired behavior.
    • 005b6ff : Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails.
    • 5e16161 : Remove CPU monitor from WebRtcVideoEngine2.
    • aef0779 : Rewrite ThreadWindows.
    • f2ec814 : Move use of DEPTH into build_with_chromium==1.
    • f88bee6 : Refactor senders into senders and sources in the simulation framework.
    • a671f4b : Fixing a VoE test to set correct rate for iSAC
    • 05db352 : Fix a bug in ACM test channel
    • 3154a1c : Reland r8210 "Add a new parameter to ACMGenericCodec constructor""
    • 4455f62 : WebRtcIsacfix_Time2SpecNeon and _Spec2TimeNeon: Fix stack alignment
    • 8820ac7 : peerconnectin_server: missing comma in sprintfn() in r8128
    • 2bbc35d : Remove unused method, SetAffinity, from the ThreadWrapper class. The method was also not consistently implemented across all platforms.
    • 6752b85 : Revert r8210 "Add a new parameter to ACMGenericCodec constructor"
    • c3643f2 : Add a new parameter to ACMGenericCodec constructor
    • 2444d96 : Control the max IPv6 Networks used by WebRTC.
    • 4ddde2e : Add arbitrary microphone geometry input to audioproc_f test utility.
    • 1398025 : Add new members to AudioEncoderOpus::Config
    • 7a37bfc : Revert 8203 "Reducing locking in OveruseFrameDetector and increa..."
    • a33f05e : Re-land "Remove (webrtc_root) from source file entries."
    • bdebccf : Fix a number of things in AudioEncoderDecoderIsac*
    • 18e7585 : Reducing locking in OveruseFrameDetector and increasing constness.
    • 50fe359 : Add tracing for slow paths in new video API.
    • 4161715 : Remove ChangeUniqueID.
    • 1ece0cb : Revert "Remove (webrtc_root) from source file entries."
    • a26f511 : Remove frame copy in ViEExternalRendererImpl::RenderFrame
    • a87c398 : Move audio_codec_speed_tests into include_tests==1 condition.
    • 2d2a1f9 : Remove (webrtc_root) from source file entries.
    • 73ca194 : Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h
    • 43c8839 : Allow rtp packet history to dynamically expand in size.
    • 827d7e8 : Change AsyncInvoker to store its closure in a scoped_refptr instead of using a raw pointer.
    • a742cb1 : Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off.
    • f17ee9c : Add case to ApmTest.Process to test the extended filter mode
    • e7a4a12 : Add arraysize() macro from Chromium, and make use of it in a few places.
    • 035e912 : Move channel_buffer.{h,cc} to common_audio.
    • a67ca1a : Only report the first rtp packet because it indicates the media has started flowing. BUG= R=juberti@webrtc.org
    • a094cac : Add stats for network merge.
    • 7d2b6a9 : Enable Clang warning implicit-fallthrough and annotate the code.
    • a907e01 : Adding constness.
    • 664ccb7 : Reland r8125: Modify some tests to never use DTX disable mode
    • 37c0559 : Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets).
    • 22c2f05 : Add "score" unit to SSIM perf score output.
    • 4aecd00 : Add support for 40 and 60 ms frames to AudioEncoderIlbc
    • 2a6558c : Make sure ByteReaderT::Read* is properly constified.
    • 7aef80c : GN: Remove webrtc_base target in favor for rtc_base.
    • 9b64a6e : Adjust parameter in videoprocessor_integrationtest for VP9.
    • dc8a9da : Adjust qp-max settinhg in VP9 wrapper.
    • 922cfcd : Use non-zero data in AudioRingBufferTest.
    • 36401ab : Update GAE API paths for join/leave.
    • 8bb32d6 : Minor updates to AudioEncoderCng
    • db1ebf6 : Add jakehilton@gmail.com to AUTHORS
    • 478cedc : Add new methods to AudioEncoder interface
    • 5614cf1 : audio_processing: Use fixed aggregation window in delay metrics
    • 6e25182 : Whitespace change after enabling gnumbd
    • ccd608e : Whitespace change for git updater
    • 0bc73a1 : Whitespace change to trigger git updater
    • f68ffca : Add PRESUBMIT check for GYP files including source files above itself.
    • 76e5e20 : Roll chromium_revision 4664fe0..9070a80 (312733:313233)
    • 273fbbb : Update StreamDataCounter with FEC bytes.
    • 70117a8 : AEC: Implements a new function for calculating delay metrics
    • fc5ad95 : Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139
    • 8501ee6 : Support VP8 HW decoding on devices with Exynos codec.
    • df9a41d : Fix bug in GetREDStatus(): it doesn't actually return the current status.
    • 82415e3 : Update AppRTCDemo to use renamed GAE messages.
    • 041035b : Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface.
    • 4dba2e9 : Consolidate anonymous namespace content and file-static methods to all be in the anonymous namespace, in preparation for refactoring a few of the functions a little.
    • d7e34e1 : Make it easier to use external libyuv + cleanup GYP files.
    • d25c034 : Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16()
    • 04cd466 : Move ThreadChecker into rtc_base_approved.
    • 38d11b8 : Enable encoder multi-threading for VP9.
    • 6f200b5 : Temporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h")
    • b6fab2b : Introduce rtc::CheckedDivExact
    • 19eb4e4 : Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h
    • 995b4c9 : Remove win_asan trybot from PRESUBMIT.py
    • acb8085 : Roll chromium_revision c086b4e..4664fe0 (312108:312733)
    • 7519de5 : Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..."
    • 0f98844 : Revert 8139 "Implement elapsed time and capture start NTP time e..."
    • dacdd94 : Reland r7980: Accept incoming pings before remote answer is set, to reduce connection latency. Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908
    • 8919cfe : Change a GYP reference to cpufeatures.gypi
    • ad3ee2c : Implement elapsed time and capture start NTP time estimation.
    • a02d768 : Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness
    • 456f014 : Re-allowing RED in voice engine.
    • 182ea46 : Remove frame copy in ViEExternalRendererImpl::RenderFrame
    • 73ee453 : Switch to use range based loops in the BWE simulation framework.
    • 36d5c3c : Leave BIO_METHOD non-const.
    • 586f2ed : Change GetStreamBySsrc to not copy StreamParams. This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple. Also, we can use lambdas now :)
    • 7e5b380 : Fix a crash in AllocationSequence. Internal bug 19074679.
    • ff108fe : Revert 8125 "Modify some tests to never use DTX disable mode"
    • b40c7bb : Change sprintf use in talk samples to snprintf
    • ea1c842 : Correct GetDriveType error handling.
    • 043db24 : Modify some tests to never use DTX disable mode
    • e5251ad : Integrate send-side BWE into simulation framework.
    • cfd82df : Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. Prepares for adding FEC bytes to the StreamDataCounter.
    • 3dd33a6 : Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes.
    • fbd37bd : Make iSAC SWB own its decoder
    • cceb166 : Fix a use-after-free when sending queued messages is aborted for blocked channel.
    • e65d9d9 : Fix an unitialized variable warning.
    • c429b82 : GN: Prepare to remove webrtc_base target
    • c78d81a : Re-land "Support 48kHz in AEC"
    • e81c5d6 : Fix TransientDetectorTest in modules_unittests on Android ARM64
    • 11af039 : Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64.
    • df7b65b : Change CreateOrGetReportBlockInformation to have one return path.
    • f938922 : Simplify and guard access to WindowsRealTimeClock.
    • 4fb7e25 : Update StatsReport and by extension StatsCollector to reduce data copying.
    • f66a6b2 : Remove unnecessary dependencies from webrtc_all target.
    • e7358ea : Only report fraction of lost packets if report_block_stats has been updated.
    • 9ffd8fe : Indentation changes.
    • fedb9ea : Correct the class name in peerconnection_jni.cc.
    • 5f93d0a : Update libjingle license statements at top of talk files for consistency
    • cbacd9e : Bump to version 41.
    • 7dba786 : Setting Opus target application.
    • 853049f : Move internal capture+render to build_with_chromium==0 condition
    • 511ab3e : Roll chromium_revision a6eafec..c086b4e
    • ee0c100 : Revert 8080 "Support 48kHz in AEC"
    • f88f88e : Remove webrtc/base/compile_assert.h
    • 9691b36 : Cleanup for Rtp Rtcp API test.
    • 8e327c4 : Update StatsCollector's interface in preparation of more changes.
    • 43e54e3 : Revert 8095 "Update StatsCollector's interface in preparation of..."
    • 5b76fd7 : Update StatsCollector's interface in preparation of more changes.
    • 474e36e : Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.
    • f9d3555 : Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test.
    • ce3ac53 : Adding TRYSERVER_PROJECT to codereview.settings.
    • 018c087 : Add /talk/examples/androidtests/{bin,gen} to .gitignore.
    • a32d154 : Disable tests failing on Android ARM64 (Nexus9).
    • ff9462e : Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan.
    • 2624b1e : Remove unused private data member engine_id_
    • fe672e3 : release the turn allocation by sending a refresh request with lifetime 0
    • d7de120 : Re-enable the messagequeue unittests. These were commented out at one point but never reenabled.
    • a1aea10 : Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps."
    • 4ba1e44 : Remove unnecessary remote bitrate estimator build rule which serves no purpose.
    • 487a444 : Add stats collection for the data channel.
    • 357469d : Fixes reference counting problem when a TransportProxy points to a Transport prior to creating channels.
    • ef2a5dd : Update AppRTCDemo UI.
    • 64d3c4b : Support 48kHz in AEC
    • 89aa276 : Fix a case where empty candidate id is used
    • d82f55d : Only adapt AGC when the desired signal is present
    • 3e42a8a : Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.
    • 32e8528 : Log configs when creating video streams in Call.
    • 1f67b53 : Remove dual stream functionality in ACM
    • 9ce01e6 : Clean unnecessary workaround for chromium import.
    • 0800db7 : Add percentage of fec packets and recovered media packets to histogram stats: - "WebRTC.Video.ReceivedFecPacketsInPercent" - "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"
    • 61c1247 : Fix a case where empty candidate id is used
    • 6c38552 : Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version.
    • 5a92b78 : Add beamforming to audioproc_float utility.
    • 6b63015 : Move ring_buffer to common_audio.
    • fd630a5 : Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior.
    • 693e01c : Fix searching for DirectX SDK during GN build.
    • f1c8b90 : Remove WebRtcVideoEncoderFactory2.
    • e5a31e1 : Revert removing of compile_assert.h.
    • 85fa94d : Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory.
    • 387841a : Improved fairness simulation by starting the flows 20 seconds apart.
    • f18fba2 : Implement SimulcastEncoderAdapter support.
    • 8315d7d : Remove dual stream functionality in VoiceEngine
    • b4e5d1b : Remove RTX SSRC when deleting the default receive stream.
    • 2ebfac5 : Remove COMPILE_ASSERT and use static_assert everywhere
    • 86e1e48 : Move system_wrappers.gyp files to the proper directory.
    • a35f741 : Add .classpath + talk/app/webrtc/androidtests to .gitignore
    • f7a5893 : Combine RegKeyTests to prevent parallel execution.
    • ef09092 : No longer asserting in mocks, split first test case in two methods.
    • 69f4738 : Roll chromium_revision 3dd2edf..a6eafec (310717:311223)
    • d6e84d9 : Always copy processed audio to output buffer in ProcessStream.
    • c0da63c : Optimize minimum delay in blocker
    • af9d56f : Unify the two copies of template_util.h
    • 0b0c241 : Only return Rtx mode in RTXSendStatus().
    • 3df38b4 : Unify the two copies of compile_assert.h
    • 58a1ba6 : Roll chromium_revision 271c6cc..3dd2edf (309333:310717)
    • 46323b3 : Remove useless AudioProcessing::Create() overload.
    • 16825b1 : Use int64_t more consistently for times, in particular for RTT values.
    • a7add19 : audio_processing: Replaced macro WEBRTC_SPL_MUL_16_16 with * in high_pass_filter
    • 2a26734 : Partial revert of r7396
    • be40eb0 : Allow 720x1280 frames encoding on Android.
    • a525c98 : Fix parallelizability in ApmTests.
    • 45db7ee : Use Java based audio as default for WebRTC.
    • 81134d0 : Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory.
    • 88a4298 : common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin()
    • c14e357 : common_audio: Made input signal const in WebRtcSplFilterMAFastQ12()
    • 19e4e8d : Add support for trying alternate server (STUN 300 error message) on TCP
    • 0ba1533 : Added support for an Origin header in STUN messages.
    • 2693a54 : Add WEBRTC_BEAMFORMER define to BUILD.gn
    • 8f27fcc : Revert 8028 "Support associated payload type when registering Rt..."
    • 80452d7 : Sync Android AppRTCDemo with internal repo.
    • 9657265 : Revert "Accept incoming pings before remote answer is set to reduce connection latency."
    • f3fd8e7 : Add NEON intrinsics version for transform_neon
    • 1592df7 : PRESUBMIT: Add GN trybots for Windows and Mac.
    • 2a16964 : Support associated payload type when registering Rtx payload type.
    • 8649fed : GN: Fix Windows build.
    • 2ead571 : Hard define the GUID for AudioEndpoint to avoid conflicts during compile.
    • 758d6d4 : audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16
    • dec649c : audio_processing/ns: Replaced WEBRTC_SPL_MUL_16_16 macro with *
    • 5e5b327 : audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc
    • 124b9c7 : Suppress races in event tracing code.
    • 5f09564 : Suppress AsyncHttpRequestTest.TestCancel leak for LSan
    • 823c9b8 : Add histograms stats for sent/received fraction loss for a stream: - "WebRTC.Video.SentPacketsLostInPercent" - "WebRTC.Video.ReceivedPacketsLostInPercent"
    • d730b28 : Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon
    • 59062d5 : Rename SendAndReceiveH264SvcQqvga to VP8 instead.
    • 8af1104 : Avoid reading past end of string in GetLine.
    • 3663fb0 : Reenable dlclose() for InternalUnloadDll on TSan.
    • bab7995 : Convert FileMediaEngineTest to use more expects.
    • 69472e7 : Add a dummy implemenation of SChannelAdapter::SetMode that makes sure that StartSSL fails if the mode is set to DTLS.
    • c10ecea : Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const.
    • dfef028 : Ignore virtual box interfaces.
    • 25dd754 : Excluding a flaky test from DrMemory
    • 7fbf278 : Suppress memcheck error in video_engine_tests
    • 1777880 : Roll gtest-parallel.
    • 07c83a1 : Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2)
    • 4e5115a : RTCPeerConnectionFactory: Explicitly create new worker and signaling threads.
    • f6a9714 : Remove peer connection and signaling calls from UI thread.
    • 2ec50f2 : Memcheck suppression for uninitalized memory in WebRtcIsac_Decode
    • d95435c : Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win
    • cbe7ca8 : Roll chromium_revision 8e72e1d..271c6cc (307131:309333)
    • 3a63a3c : iOS AppRTC: First unit test.
    • 4796cb9 : Disable flaky RelayServerTest.TestExpiration on all platforms.
    • fb7a039 : Use array geometry in Beamformer
    • a37bf2c : Hack clock_unittest fix for parallel execution.
    • c37e72e : Make setting identical RTP extensions a no-op.
    • e5a921a : Use tmp files in file_utils_unittests
    • 76bc981 : Use a temp file in FileLockTest.
    • 433006a : Fixed style issues from lint and got rid of unused fields.
    • c4ad157 : Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9.
    • 215bbbd : Fix for log typo in ViEExternalCodecImpl::RegisterExternalReceiveCodec.
    • aeb0dd3 : Disable RelayServerTest.TestExpiration on Mac.
    • 8390c27 : Add two unit tests for Android AppRTCDemo.
    • 896888b : Remove min bitrate from simulcast streams.
    • bac0012 : Extend delay estimation window in AEC to 500 ms on all platforms
    • 9eacb8c : Make P2PTestConductor use VirtualSocketServer.
    • c62749f : Parallelize MediaRecorder unittests.
    • 3a70625 : audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877
    • 27f5317 : Use the prod GAE server in AppRTCDemo for iOS.
    • 5eb71eb : Fix style issues from lint.
    • 34ac956 : Do not use openmax_dl for MIPS64 platform.
    • b2bda67 : Removing old channel code from a few more places.
    • a9b1ec0 : Support for DTLS in OpenSSLAdapter
    • c5fd66d : Accept incoming pings before remote answer is set to reduce connection latency.
    • 84d8447 : Minor fixes regarding accumulator usage on MIPS platforms.
    • b024da3 : Add support for audio device selection in AppRTCDemo.
    • 5ad4178 : Move the Jingle-specific network code into webrtc/libjingle.
    • 46d4d29 : Add field trial for screenshare bitrates when using temporal layers.
    • 1be0a78 : Removing giles@mozilla.com from WebRTC watchlist.
    • 53cb741 : Make RelayServerTest use VirtualSocketServer.
    • 086c8d5 : Use a temporary buffer to scale a screencast in OnFrameCaptured
    • 4c0544a : Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository.
    • ed1a48b : Fix mac video capture leak.
    • 7ce4a58 : Add initWithCoder to RTCEAGLVideoView.
    • ae643ce : Wire up Beamformer in AudioProcessing
    • 8817256 : Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator.
    • 50f7db8 : Remove unneccessary lock causing a potential deadlock.
    • a6f7ba6 : Add a AppRTCDemo setting to change the GAE server.
    • 5570769 : Remove the last getters from VideoReceiveStream stats.
    • 742386a : Enable payload-based padding by default and remove the API.
    • aa21f27 : Unify the two copies of move.h
    • d16e839 : Rtp-Rtcp sender cleanup.
    • 556caff : GN: Fix build for Mac
    • 11d8176 : Move updating nack bitrate inside UpdateNACKBitRate.
    • 5647877 : Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
    • 1090a6e : Remove obsolete target_arch == armv7.
    • aacc234 : Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
    • 16a05dd : Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode.
    • f5847d7 : Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well.
    • cb79141 : Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc. When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap.
    • ce4e9a3 : Refactor some receive-side stats.
    • 98c04b3 : Get avg_delay_ms from DecoderTiming callback.
    • 9b79197 : Suppress REMB in bitrate ctrl if it seems lika a short network glitch.
    • f832a6d : Remove _t from function pointer typedefs.
    • eed7a22 : Make an AudioEncoder subclass for iSAC redundant encoding
    • dd8f6f3 : Rename rtpDumpPktHdr_t to RtpDumpPacketHeader.
    • a9cf079 : Rename external_hmac_ctx_t to ExternalHmacContext.
    • e468bc9 : Rename _t struct types in audio_processing.
    • cab1291 : Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder
    • 4fba293 : Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port
    • 4cb3856 : Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
    • 536f999 : Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
    • c51fb93 : Fix an assert failure caused by race condition
    • 0ab42bc : Make safe_conversions suitable for rtc_base_approved.
    • bc03192 : Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository.
    • 0eb6eec : Move VirtualSocket into the .h file to allow unit tests more control over behavior.
    • 6f10ae2 : Support block_size greater than chunk_size in Blocker
    • eb54446 : Rename _t struct types in audio_coding.
    • 209df9b : Change MockStatsObserver to grab values inside of OnComplete. This is done since StatsReportCopyable is going away and the list of supported properties of the mock class is known. StatsReports holds a list of pointers to objects that cannot be cached, so this is a simple way to grab the values when they're available.
    • e728ee0 : Remove or rename typedefs with _t prefixes.
    • 5263c58 : Add a little utility to capture cpu graphs.
    • 70f74f3 : Add overshoot of target bitrate for screenshare with temporal layers.
    • 45a272a : Change aggregated fraction loss to be calculated from the cumulative loss and extended sequence number diff between the current and the last report block of two get stats calls.
    • e102e81 : Enable the iSACfix AudioDecoder test (and make it work again)
    • 38881be : If one of the bundled content is missing in SDP, return false to MaybeEnalbeMuxingSupport(). Verified in chromium. Now the existing content still could work.
    • 950c518 : Add adapter_type into Candidate object.
    • 971bf55 : Fix path to mock_agc.h
    • f050791 : Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
    • 4afb599 : Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
    • e2b7585 : Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository.
    • a32487f : Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder
    • 02c21db : Make one OWNERS files for all of webrtc/libjingle so we don't need approval from webrtc/OWNERS every time we want to add a directory.
    • 08df9b2 : Add a manageable command-line tool for AudioProcessing.
    • cf6d0b6 : Add 48kHz support to AGC
    • 2510d11 : Add (safe) uint32_t cast to fix Win64 build.
    • 048c502 : Handle all permissible PCM fields with WavReader.
    • 451a133 : Add AGC manager tests.
    • c1c9291 : Make an AudioEncoder subclass for RED
    • 88bdec8 : AudioEncoder subclass for iSACfix
    • 0198933 : Cleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate().
    • d08d389 : Add field to counters for when first rtp/rtcp packet is sent/received. Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).
    • b395a5e : audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/
    • 55360ae : Revert "Add adapter_type into Candidate object."
    • d021bbb : Fix vp9 setting in vie loopback test.
    • aaf02cc : Add adapter_type into Candidate object.
    • 0b1534c : Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
    • 96a6262 : Remove 20ms support in AGC
    • 1f05c45 : Reenable test case P2PTransportChannelTest.TestIPv6Connections
    • 903b4ae : Removes unused test files by audio_processing/transient
    • dd32213 : resources/audio_processing: Removed unused test files
    • 6fd9308 : Suppressing warnings in GetRTT() in VoE.
    • e2e199b : Clean up StatsObserver's OnComplete methods (address TODOs).
    • 3440fe1 : Use webrtc_root instead of DEPTH for iSAC.
    • 032b802 : (Auto)update libjingle 82121498- 82126219
    • dd0601f : Remove unneeded ctor and add a more practical one The default constructor isn't necessary, so I'm removing it. I'm adding another one so that we can (later) make |type| const.
    • 69bc5a3 : Add thread asserts to StatsCollector. Also adding a "ForTest" postfix to a test-only method.
    • fb108b5 : Revert r7885.
    • b413a30 : Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.
    • 18a3896 : Revert r7886:7887.
    • 40e4767 : Add NEON intrinsics version for min_max_operations_neon.c
    • e575e9c : Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h
    • e9db7fe : Put pseudotcp back because remoting uses it.
    • dee76f3 : Move the obvious/easy Jingle-specific code into webrtc/libjingle.
    • 8c9d79a : Add adapter_type into Candidate object.
    • c57310b : Switch kStatsValueName* constants to be enums instead of char*. This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.
    • 3b79daf : Moving encoded_bytes into EncodedInfo
    • c8bc717 : Fix webrtc gn windows build.
    • f68faa5 : Removing manual test pages because they have been moved to github.
    • 40b276e : Cleanup little things found when refactoring.
    • 27d106b : Move the downmixing out of AudioBuffer
    • 0ca768b : Adding DTX to WebRTC Opus wrapper (relanding).
    • 2b19f06 : Wire up RTT statistics to webrtc::Call.
    • 1351895 : Remove old_factory from WebRtcVideoEngine.
    • 128faba : Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin...""
    • 626c09f : Move isolate path into webrtc/build/android/test_runner.py
    • 817e50d : Make an AudioEncoder subclass for PCM16B
    • b3ad8cf : Make an AudioEncoder subclass for iSAC
    • abe3f18 : Checking whether ACM uses codec internal or WebRTC DTX.
    • 55d42c3 : DCHECK: Reference condition parameter in release builds
    • cd5b209 : Deleting quality dashboard code.
    • 3c31e6e : Add NEON intrinsics version for WebRtcSpl_MinValueW16Neon
    • f4c1948 : Remove jitter_estimate_test.h
    • c5ebbd9 : Support 48kHz in Noise Suppression
    • d8ca723 : Remove CELT support from audio_coding.
    • 8084f95 : Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval.
    • 85bd53e : Add AbsSendTime unittests to rampup_tests.cc.
    • 0df3715 : Cast payload type to int in logs.
    • a853077 : (Auto)update libjingle 81702493- 81755413
    • 3cd26b6 : Revert r7858 ("DCHECK: Reference condition parameter in release builds")
    • 3148060 : DCHECK: Reference condition parameter in release builds
    • ff1a3e3 : Make an AudioEncoder subclass for comfort noise
    • 6fd52f3 : Add NEON intrinsics version for WebRtcSpl_DownsampleFastNeon.
    • ae20d3b : Add NEON intrinsics version for WebRtcSpl_CrossCorrelationNeon.
    • aa2c342 : Add back a constructor to fix FYI build.
    • 5c32a84 : Attempt to fix FYI bots. The FYI bots went red after https://webrtc-codereview.appspot.com/32179004/ landed.
    • 87776a8 : iAppRTCDemo: WebSocket based signaling.
    • 0babb4a : Fix a comment.
    • c9d155f : Move implementation of types in statstypes. to its cc file.
    • a954c07 : AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
    • 19dd129 : Revert 7846 "Adding DTX to WebRTC Opus wrapper"
    • f244760 : Add histograms for receive statistics: - decoded frames per second ("WebRTC.Video.DecodedFramesPerSecond") - percentage of delayed frames to rendered ("WebRTC.Video.DelayedFramesToRenderer") - average delay (of delayed frames) to renderer ("WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs")
    • 4321f17 : Adding DTX to WebRTC Opus wrapper
    • 5c3ee4b : Add empty implementation file that will hold statstypes.h implementation. The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc.
    • 1784d7c : Adding an codec interal CNG test in NetEq.
    • e04a93b : Move the AudioDecoder interface out of NetEq
    • 97d0489 : Add video send bitrates to histogram stats: - total bitrate ("WebRTC.Video.BitrateSentInKbps") - media bitrate ("WebRTC.Video.MediaBitrateSentInKbps") - rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps") - padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps") - retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")
    • 7ba9f27 : Set CHECKOUT_SOURCE_ROOT environment variable for Android test wrapper.
    • eef8538 : Fix AppRTCDemo closing error for KK and JB Android devices.
    • 86b6d65 : Remove no longer used video codec test framework.
    • 8911bc5 : Add AudioEncoder::Max10MsFramesInAPacket
    • 130fef8 : Bugfix in AudioDecoderTest
    • edeea91 : Change all system clock types to int64_t in bitrate_controller.
    • fcbe36a : Add const qualifier to WebRtcPcm16b_Encode
    • a1ef7bf : ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable.
    • 3b3c406 : Revert 7826 "Change Android PeerConnectionUnittest to build usin..."
    • cb858ba : Make an AudioEncoder subclass for iLBC
    • ee43263 : Cleaned up real_fft APIs due to non-existing NEON code
    • ed7824b : Change Android PeerConnectionUnittest to build using Chrome macros. The purpose is to be able to run the tests using Chromes buildbots. To run: CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
    • ba8138b : Change type of nack_last_time_sent_full_ from uint32_t to int64_t. Could cause nack requests to be sent too frequently.
    • aefe61a : PRESUBMIT: Add check for checkdeps.
    • 7db359b : Roll chromium_revision 24b4c73..8e72e1d
    • d91d359 : PRESUBMIT: Add iOS ARM64 trybots to default set.
    • fb01376 : Adjust some parameters for VP9 tests.
    • e2a9261 : Improve AppRTCDemo connection speed by sending all http POST requests asynchronously.
    • bd8cc0b : Add codereview.settings to the /talk subdirectory
    • 5af8cd7 : Add codereview.settings to the /webrtc subdirectory
    • 599e299 : cricket::VideoFrame int64 to int64_t.
    • 9b5467e : Fix assertion failure when closing data channel, and add a unit test.
    • 4b407aa : Update AppRTCDemo README with information on 3-dot-apprtc server and new command line arguments.
    • 7169afd : With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior.
    • 369746b : Support new WebSocket signaling format.
    • 0b38478 : Add support for parsing header only RTP dumps with bwe_rtp_play.
    • 33ccdfa : Relanding r7807.
    • 52bc4f4 : Revert 7807 "Removing unused opus wrapper APIs."
    • c0991fe : Roll chromium_revision 24b4c73..f27c369
    • e54a634 : Removing unused opus wrapper APIs.
    • 8c9ff20 : Redo the change of https://webrtc-codereview.appspot.com/30949004/
    • fd84229 : Revert "Implement GetState() for channel's connectivity check state."
    • ff72f9e : Implement GetState() for channel's connectivity check state.
    • fd4acf6 : Adding WebRtcSpl_MaxAbsValueW16 intrinsics version
    • 3a52458 : add WebRtcIsacfix_AutocorrNeon's intrinsics version
    • 8dc21dc : Rename internal AudioEncoder::Encode method to EncodeInternal
    • d1fac61 : Remove need for assembly offset generation in aecm and ns module.
    • 3800e13 : Revert r7798 ("Move the AudioDecoder interface out of NetEq")
    • 00ba1a7 : Move the AudioDecoder interface out of NetEq
    • 0fb6ad2 : Check if cpu_monitor_ exists before Stop().
    • fa914e2 : Adding a duration printout to neteq_rtpplay
    • d8aed6b : Verify that cpu_monitor exists before calling Stop().
    • c3e097c : Add Android test runner script for WebRTC.
    • 8e5c814 : Convert DEPS to only reference Git repos
    • 511f8a8 : TurnPort should ignore STUN binding reponses when using shared socket.
    • 001f3b9 : Adjust parameter in videoprocessor_integration_test for vp9.
    • a7384a1 : Simplify audio_buffer APIs
    • ceca014 : Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9.
    • eb09542 : Don't reset sequence number for a stream on deactivate/reactivate.
    • d019551 : Change minimum video encoder initialization resolution to 176x144 to ensure HW encoder can be initialized.
    • 1751ee7 : Remove -flax-vector-conversions flag for ARM NEON building.
    • ac68ef9 : Clear 2 unused functions in audio processing aecm module.
    • beee9ce : Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video. The reason is that the desktop apprtcdemo only handle one MediaStream and this doesn't play audio if it receive two streams.
    • 7f1dfa5 : Adding a payload type to AudioEncoder objects
    • 0cd5558 : AudioEncoder subclass for G722
    • 84515f8 : Roll chromium_revision 309cf65..24b4c73
    • 5950b64 : Use c++11 features in webrtc/base/network.cc as a test to see if we can use them.
    • 146e0fd : Fix the build by putting in a typecast to avoid a comparison between signed and unsigned ints introduced in cl/81073932.
    • dea5173 : Add start bitrate and vp8 hw acceleration option to Android AppRTCDemo.
    • 32ec0dd : (Auto)update libjingle 81063831- 81073932
    • 7f72249 : Set simulcastIdx field to zero even if it has no meaning. Helps to be able to memcmp between 2 parses of the same packet.
    • 273a414 : Report encoded frame size in VideoSendStream.
    • 1db20a4 : Adding EncodedInfo struct to AudioEncoder::Encode
    • 20446e7 : Move and rename neteq/test/RTPcat to neteq/tools/rtpcat
    • c93437e : Add test NetEqDecodingTest.CngFirst
    • 8331714 : Adding a new test helper RtpFileWriter and use it in RTPcat
    • 4796301 : Whitespace change to force builds.
    • e75f2ce : Add FORCE_HTTPS_COMMIT_URL to codereview.settings.
    • cc7755b : Whitespace change
    • 74499ef : Add whitespace.txt file.
    • 2c13f65 : Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'.
    • 83b5200 : Add framerate for complete received frames to histogram stats: "WebRTC.Video.CompleteFramesReceivedPerSecond".
    • cc144de : Make bands vector in SplittingFilter Analysis const
    • 8789376 : Move ChannelBuffer class to channel_buffer file
    • d87213a : Remove unused RtpStatistics struct.
    • 7d4e6d0 : Roll chromium_revision d8c9041..309cf65
    • d952c40 : Add receive bitrates to histogram stats: - total bitrate ("WebRTC.Video.BitrateReceivedInKbps") - media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps") - rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps") - padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps")
    • 3e9ad26 : Refactor iOS AppRTC parsing code.
    • 79b9eba : Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands
    • 7806d8f : Fix an ASSERT that fires in a browser test for renegotiation. See https://code.google.com/p/chromium/issues/detail?id=293125#c33
    • a71bb60 : Revert 7750 "Don't reset sequence number for a stream on deactiv..."
    • a56a2c5 : Enabling building with NEON on ARM64
    • 31f7a0e : Don't reset sequence number for a stream on deactivate/reactivate.
    • 91d928e : Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader
    • 2faf7ee : Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection.""
    • 58edb83 : Add video encoder fps and bitrate statistics to Android AppRTCDemo UI.
    • 0087318 : Implement settable min/start/max bitrates in Call.
    • b951eb1 : Add back EXPECT_TRUEs.
    • ba25347 : Reenable GetStats test.
    • dab5d92 : Use mirror image for Android AppRTCDemo local preview.
    • 03499a0 : Add wav output capability to neteq_rtpplay
    • aff1751 : Add new test for VP8 packetizer to test tight partitions
    • dde19a6 : sync_chromium.py: Check for chromium/src
    • 3398a4a : PRESUBMIT: Only notify GN changes for GYP files in webrtc/*
    • 8562f23 : OWNERS: Remove tomasl@ and mallinath@
    • 4f16c87 : Simplifying VideoReceiver and JitterBuffer.
    • 9334ac2 : Use vector of CSRCs for DeliverFrame & SetCSRCs.
    • 308e7ff : Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."
    • 2751f2a : This adds an Android apk for running tests on the Java layer of PeerConnection.
    • 88d14f4 : Remove expensive and unnecessary memory alloc for sending black frames on video mute.
    • 1153322 : Build fix for MIPS Android Webview build.
    • bdcf38c : cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class
    • ad0e71c : Update mock_frame_dropper.h to use size_t
    • 4591fbd : Use size_t more consistently for packet/payload lengths.
    • edc6e57 : Support loopback mode and command line execution for Android AppRTCDemo when using WebSocket signaling.
    • 6ff3ac1 : Fix problems if first packet into NetEq is rejected
    • ed91068 : Create a NetEq test for when the first incoming payload type is unknown
    • 049e4ec : Change default values for CpuOveruseOptions. Enabled method based on encode time and modified values for the low (60-55) and high threshold (90-85).
    • f58b455 : cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
    • 40af3a5 : Revert "Add DCHECK to ensure that NetEq's packet buffer is not empty"
    • 6f6ef72 : Add DCHECK to ensure that NetEq's packet buffer is not empty
    • 2176db3 : AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land)
    • c56814f : Roll chromium_revision 91f1781..d8c9041
    • 087da13 : Add empty 3 band splitting filter API
    • 2656bf8 : Fix ExpectedQueueTimeMs() to avoid truncation or overflow.
    • 930e004 : Add jmi field for packets discarded due to network error
    • c72a22c : Add preliminary empty file videoframefactory.cc
    • f5b56fb : Annotate COMPILE_ASSERT with __attribute__((unused)).
    • 4ef22d1 : Setting Opus FEC as default
    • 966a708 : Use RtpFileSource in NetEqDecodingTest
    • 4ec19e3 : Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..."
    • 858dbbc : cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
    • 6a782c2 : Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases.
    • be05c74 : Wrap the splitting filter in its own class
    • 67c2247 : Disable EndToEnd.GetStats test.
    • a73d746 : Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..."
    • bbd8cad : cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
    • ece3890 : Report total bitrate for all streams in GetStats.
    • 35c1ace : Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..."
    • a1f5b96 : Remove unnecessary copying of libjingle resource files.
    • 52da44b : WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
    • 49ff40e : Make SetREMBData accept vector of SSRCs.
    • a9c2d45 : Fix and enable CanReceiveFec test.
    • ee30082 : Set correct sample rate in far_frame in audioproc tool.
    • 52bb521 : Update isolate files for Android APK tests.
    • 312614a : Add jmi field for packets discarded due to network error
    • 90b9b08 : Fix a platform check to use WEBRTC_WIN instead of OS_WIN.
    • 6ca6190 : Fix a SCTP message reordering issue in datachannel.cc. Previously DataChannel::SendQueuedDataMessages continues the loop of sending queued messages if the channel is blocked, which will cause message reordering if the channel becomes unblocked during the loop, i.e. messages attempted after the unblocking will be sent earlier than the older messages attempted before the unblocking.
    • ea73ff7 : webrtc::Scaler: Preserve aspect ratio
    • 0b3d89b : VideoSendStreamTest.SwapsI420VideoFrames: Initialize frame memory to avoid drmemory errors
    • 14ea50a : Change the static_library("webrtc") to a source set in the GN build.
    • 0e37b89 : replace inline assembly WebRtcAecm_CalcLinearEnergiesNeon by intrinsics.
    • e497be3 : replace inline assembly WebRtcAecm_StoreAdaptiveChannelNeon by intrinsics.
    • 0e71070 : Use ScreenCapturer to capture the whole and clip to the window rect when the shared window is on the top.
    • a367aea : Bump to version 40
    • f7c5d4f : Revert 7679 "webrtc::Scaler: Preserve aspect ratio"
    • 525baea : Add PROJECT to codereview.settings
    • 944fb57 : Roll chromium_revision 375f736..91f1781
    • 809986b : webrtc::Scaler: Preserve aspect ratio
    • cd621a8 : Add thread annotations to overuse_frame_detector class.
    • 8038d42 : Follow-up fixes for G722
    • 1431e4d : Revert 7675 "Make an AudioEncoder subclass for iSAC"
    • 05feff0 : Make an AudioEncoder subclass for iSAC
    • 33045ab : Change from talk/p2p (r7664) "(Auto)update libjingle 79414100- 79428003".
    • 43e033e : Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted."
    • 4ffc734 : replace inline assembly WebRtcAecm_ResetAdaptiveChannelNeon by intrinsics.
    • d024f75 : clear asm code and unused functions in audio processing module
    • c492231 : Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds.
    • d819803 : Wire up DSCP support in WebRtcVideoEngine2.
    • 83d4804 : Put send-side bwe probing under finch experiment.
    • 957e802 : Refactor SetDefaultEncoderConfig to work on existing codecs.
    • a5d29fc : Add unit to dropped frames.
    • bd495fa : .gitignore updates
    • 3c1970f : (Auto)update libjingle 79414100- 79428003
    • 188d3b2 : Enable VP9 video codec support on webrtcvideoengine behind a field trial.
    • f85dbce : Reapply "Advertise G722 as 8 kHz rather than 16 kHz""
    • d105cc8 : Change dummy address to use 0.0.0.0 instead of :: This is to not break compatiblity with FF.
    • d42a3ad : Remove partially defined WebRtcRTPHeader from Parse().
    • a2ef4fe : Prevent a lot of VideoSendStream reconfigures.
    • 82775b1 : Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime. This will allow to plugin VP9 based on a field trial.
    • 5e16066 : Reland Volume buttons in AppRTCDemo should affect output audio volume (part I).
    • 332331f : Use uint16s for port numbers in webrtc/p2p/base.
    • d89b69a : Fix WebRTC Win64 + BoringSSL build.
    • dd43bbe : Volume buttons in AppRTCDemo should affect output audio volume (part II).
    • dced5d7 : Revert "Advertise G722 as 8 kHz rather than 16 kHz"
    • 34bda43 : (Auto)update libjingle 79326895- 79329222
    • e5421e9 : Volume buttons in AppRTCDemo should affect output audio volume.
    • fd0efb6 : Remove deprecated PeerConnection APIs. Removes PeerConnectionObserver::OnError. Removes MediaConstraints argument to PeerConnection::AddStream. None of these have ever been implemented and have been removed from the spec.
    • 19b4741 : Removing unused method GetDefaultVideoEncoderConfig.
    • 931e3da : Log formatting fix for VideoEncoderConfig.
    • 0ef890a : (Auto)update libjingle 79285346- 79320771
    • 6340acd : AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation.
    • 1dcca40 : Advertise G722 as 8 kHz rather than 16 kHz
    • 8b2058e : Remove the state_ member from AudioDecoder
    • 32022c6 : Revert 7642 "Fix memcheck and dr memory after flakiness dashboar..."
    • 724fbaf : Fix memcheck and dr memory after flakiness dashboard deployment.
    • 7e4a05e : Exclude SendsAndReceivesVP9 for linux-memcheck.
    • 53bed75 : Change DrMemory exclusion to match changed test name.
    • f6b7c7e : Exclude SendsAndReceivesVP9 for WinDrMemory.
    • e1745cb : Adjust parameter in vp9 rate control test.
    • 5f1e2e4 : Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.
    • ee9d61c : This fixes a small memory leak (found using Xcode/Instruments on iOS) in the ObjC bindings of PeerConnection. The generated session description has to be released by the recipient
    • 6a364fe : Remove uses of build date/time.
    • 0bae1fa : Wire up bandwidth stats to the new API and webrtcvideoengine2.
    • a22a628 : (Auto)update libjingle 79205306- 79244016
    • 72fd339 : Restore old behavior for Android in fileutils.cc
    • f6e1600 : Roll chromium_revision d3db2ff..375f736
    • dc86624 : Fix android_clang build.
    • 368215d : Revert 7623 "Remove the state_ member from AudioDecoder"
    • 8a232f6 : Revert 7625 "Don't use DCHECK when you need the side effects..."
    • 795d003 : (Auto)update libjingle 79200114- 79205306
    • 8125744 : Cleanup RTCVideoRenderer interface.
    • b8425bc : Don't use DCHECK when you need the side effects...
    • 45ecf4c : (Auto)update libjingle 79169148- 79192489
    • 9e52558 : Remove the state_ member from AudioDecoder
    • 7c29e8c : Add support for VP9 in webrtc::Call and video_loopback.
    • d839e0a : Reduce to 2 probes when probing for initial bandwidth.
    • db26247 : Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged.
    • 8944c9d : AppRTCDemoActivity: use differnet Themes for different API levels
    • d367321 : Add kjellander as PRESUBMIT.py OWNER
    • dcebf2d : Reworked paced sender queue
    • fad9aec : Remove protected files from talk/PRESUBMIT.py.
    • 88ef632 : Falling back on single-stream on multiple SSRC.
    • 28af641 : Presubmit was not whitelisting libjingle_tests.gyp or sound.gyp due to a missing comma leading to a concatenation of the two strings in the whitelist.
    • b3265ac : Adds support for finch experiments to video_loopback.
    • 52b42cb : Fix problem with late packets in NetEq
    • 09cc686 : Delete VideoReceiveStream channels in destructor.
    • 6de75ca : Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16
    • c78cf97 : Remove the useless dummy state parameter to WebRtcG711_*
    • b5d045e : ReAdd PeerConnectionInterface::AddStream to fix Chrome build. AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints); This will be removed once Chrome has been updated.
    • 18de6f9 : Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send. The problem with Thread::Send is that it processes incoming pending messages and for the proxies, this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior.
    • 721ef63 : Remove the codec_type_ member from AudioDecoder
    • c2dd5ee : Prepare for removal of PeerConnectionObserver::OnError. Prepare for removal of constraints to PeerConnection::AddStream.
    • f37145f : Enables AIMD control by default.
    • b0f4b3d : Improving error message from neteq_rtpplay
    • a663d90 : (Auto)update libjingle 79104430- 79104922
    • 5f38c8d : Android AppRTCDemo improvements: - Add a room list to ConnectActivity with buttons to add/remove rooms. - Add loopback call button. - Add option to toggle full screen / letterbox video. - Add camera fps settings. - Fix device to landscape orientation for HD video until issue 3936 will be fixed. - Fix a few crashes by avoiding calling peer connection and GAE signaling function while connection is closing. - Better handling GAE channel error - catch channel exceptions and display dialog with error messages.
    • 5804936 : Add format members to AudioConverter for DCHECKing.
    • e451b75 : Update rate control parameter in vp9 test.
    • 4765ca5 : Roll chromium_revision: 28d1981..d3db2ff
    • f866b2d : Restore the void return type on WriteWavHeader.
    • b81e304 : replace inline assembly WebRtcNsx_AnalysisUpdate by intrinsics.
    • f947180 : Add Opus support to neteq_rtpplay
    • 96a9325 : Implement external decoder support in WebRtcVideoEngine2.
    • 548b228 : Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call.
    • 96dc685 : Add stats for video: - number of sent/received RTCP NACK/FIR/PLI per minute - percentage of unique sent/received NACK requests - percentage of discarded/duplicated packets by the jitter buffer - permille of sent/received key frames
    • 2236267 : Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan
    • bf09976 : Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already.
    • ed45896 : Adjust/increase rate control thresold for a vp9 test.
    • 5b88317 : Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test.
    • 5072e0f : Update Android projects to API level 21.
    • 818c9f9 : replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics.
    • a3ed713 : Add a WavReader counterpart to WavWriter.
    • c2c94a9 : Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64
    • 78c222b : Update all .isolate files for the new format.
    • 8a130c1 : Update Android projects to API level 20.
    • 053c6ab : Fix N7 camera aspect ratio.
    • 508c916 : Build fix for MIPS32R6.
    • cc476aa : Fix a name collision with Android libc++
    • b7ed779 : Implement conference-mode temporal-layer screencast.
    • 3bf3d23 : Configure A/V sync in WebRtcVideoEngine2.
    • 4abadab : Simplify bwe tests.
    • 2dc6f31 : Adapting bitrate according to maxplaybackrate for Opus.
    • 8328e7c : Revert "Revert part of r7561, "Refactor audio conversion functions.""
    • 14146e4 : arm64 iOS build.
    • 50ca986 : Improve the logging when a TCP connection is deleted.
    • d0cf68e : Add 15 fps support for Android devices with missing 15 fps camera mode.
    • 8aa4d2d : Creating a C++ wrapper class for VAD
    • bcfb4d0 : Revert part of r7561, "Refactor audio conversion functions."
    • 8219529 : Cleaning up r7562-7567.
    • 879fac8 : (Auto)update libjingle 78822708- 78823675
    • 5f73a37 : Revert 7563 "before rebase" due to wrong submission
    • c11cc8d : Revert 7564 "to submit" due to wrong submission
    • de386bf : to submit
    • c673bb9 : before rebase
    • 0b62672 : adding default rates
    • 4fc4add : Refactor audio conversion functions.
    • 776e6f2 : Use external VideoDecoders in VideoReceiveStream.
    • 2dd3134 : Add stats for duplicate sent and received NACK requests.
    • f567095 : common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32
    • 7f10513 : Remove unused code in overuse detector.
    • decd930 : AudioEncoder: num_10ms_frames_per_packet - Num10MsFramesInNextPacket
    • cfe3845 : Enable G.722 for Chromium builds
    • 1abc146 : (Auto)update libjingle 78738075- 78738103
    • 7998089 : ApprtDemo Android: Switch between front and back camera. This adds a UI icon for switching between the front and back camera. This cl adds the possibility to change between the front and back camera while in a call or before the other end have connected.
    • 663fdd0 : Make an AudioEncoder subclass for Opus
    • 2623695 : Renaming bandwidth to bitrate in webrtcvoiceengine.
    • ffeaeed : Make NSinst_t* const and rename to self in ns_core
    • 269fb4b : move xmpp and p2p to webrtc Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.
    • 8b1b23f : Make local functions static and dropWebRtcNs_ in ns_core
    • 28b5467 : Make all comments whole sentences in ns_core
    • bd6bdca : scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots.
    • ae694ef : (Auto)update libjingle 78642371- 78680406
    • a296725 : audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with "
    • 67ca26e : common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16
    • ff8a98e : Use neteq_unittest_tools in audio_decoder_unittests
    • 820efd5 : Fix double backslashes in incoming_video_stream.cc
    • fbd55cb : (Auto)update libjingle 78616359- 78642371
    • f15dee6 : Check if a datachannel in the current local description is an sctp channel before assuming rtp. When generating an offer from a local description when 'sctp' is not explicitly set in the media session options, we were generating an offer with an RTP datachannel even though the channel in the local description was already sctp.
    • aada86b : Add a simple AudioConverter class.
    • 33a0e2d : Only configure the SSL library in one place.
    • aca5803 : Move (test) RtpFileReader to a lightweight target.
    • b787f4c : Move scoped_ptr "free" functions into the webrtc namespace.
    • 243eb8e : Adding setting screen to AppRTCDemo.
    • 068b529 : (Auto)update libjingle 78583324- 78583691
    • df42988 : Upgrade our scoped_ptr copy to match Chromium's latest.
    • 2e7ee4b : Fix the SrtpFilter crash caused by two local offers.
    • efc82c2 : Implement screencast settings for WebRtcVideoEngine2.
    • a37f1dd : Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile
    • 0552356 : isacfix: Refactor big-endian reading and writing
    • 9fed099 : Increase max trace message size to 1024 characters.
    • c86ec3e : Fix ::~LogMessage to print as a string.
    • 1732df6 : Use flags set by the port allocator.
    • 3b839d0 : PRESUBMIT: Add linux_msan to default trybots.
    • 3f7bcc1 : (Auto)update libjingle 78430441- 78445452
    • c7ed8db : (Auto)update libjingle 78427027- 78430441
    • 4709887 : Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected.
    • 39b1743 : Adding the subtool rtcBot report visualizer
    • ad3b5a5 : Move min transmit bitrate to VideoEncoderConfig.
    • c9d6d14 : patch from issue 25469004
    • 8fe75ee : (Auto)update libjingle 78381351- 78389679
    • fb5e9fc : (Auto)update libjingle 78344087- 78381351
    • 7e19a11 : Break out WebRtcNs_ComputeDdUpdate function in ns_core
    • f8ea0d5 : Break out WebRtcNs_UpdateNoise function in ns_core
    • 799e88a : Break out FFT function in ns_core
    • 8454ad8 : Break out ComputeSnr function in ns_core
    • 0d3e254 : Adding three video conference bots test
    • 0e19d0c : Adding file from test.webrtc.org domain to be downloaded
    • 580d367 : Add macros and APIs for webrtc histograms.
    • 9d446f2 : (Auto)update libjingle 78296920- 78342456
    • 8539bd0 : Download full Chromium checkouts by default
    • 82462aa : Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.
    • 2192701 : Using the Unused turn configuration in two way test
    • ad553a2 : Let video_loopback use internal VCM capturers.
    • 15c717b : Add a memcheck exclusion for EndToEndTest.CanSwitchToUseAllSsrcs.
    • a9f0898 : (Auto)update libjingle 78273470- 78296920
    • 7bb4a98 : Merging Henrik's and Peter's changes for AppRTCDemo from https://github.com/hkjellander/AppRTCDemo.
    • fce8f5d : NOTE: This code review based on the running issue: https://webrtc-codereview.appspot.com/24939004/
    • 3382059 : Adding Two way video and audio streaming test to RtcBot
    • e9b7d03 : HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test.
    • fb5410a : (Auto)update libjingle 78262388- 78262615
    • eacc6e4 : Remove some disabled tests in WebRtcVideoEngine2.
    • 82e430c : Suppress libyuv uninitialized read in CopyRow_AVX
    • 32452b2 : Make ReconfigureVideoEncoder use current bitrate.
    • 860ccc9 : Tighten up MSan blacklist.txt owners.
    • 3f8f555 : Disable TestVp8Impl.BaseUnitTest on MSan.
    • 76960d5 : For FIR packet, payload length is zero, so SendToNetwork function is failing.
    • 1d9af96 : Roll chromium_revision de13cf4..28d1981 (299488:300483)
    • 67cf1d7 : Break out WebRtcNs_Windowing function in ns_core
    • 0e70992 : Break out WebRtcNs_Energy function in ns_core
    • 7634c09 : Break out WebRtcNs_IFFT function in ns_core
    • a5c36b3 : (Auto)update libjingle 78193292- 78199328
    • b6173ab : Fix local address leakage when IceTransportsType is relay
    • 333e255 : Break out WebRtcNs_UpdateBuffer function in ns_core
    • 1288cbb : (Auto)update libjingle 78106439- 78193292
    • def1e97 : Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests
    • 78ea06d : audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with
    • 913f7b8 : Fix for glitches in ACM when switching desired output sample rate
    • a8c0edd : Avoid using EGLContext class for Android 4.1 and below.
    • b69ea9a : common_audio: Replaced invalid operand in min_max_operations_neon.S"
    • fa553ef : Set up start bitrate in WebRtcVideoEngine2.
    • b35b136 : Make avg_{psnr,ssim}_threshold_ const.
    • 2abebe7 : audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with
    • a5ce7bb : audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with
    • 28100cb : Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
    • 7992b40 : (Auto)update libjingle 77953038- 77970462
    • b1dac33 : Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
    • 5820294 : Cleaning up Android AppRTCDemo.
    • 0371a37 : Moving creating TURN configration to the host machine instead of the bots - rtcBot
    • f7030d4 : Query Android device orientation on every camera frame received.
    • 9c58ea8 : rtc_unittest: copied gtest excludes from libjingle_p2p_unittest since its tests have move to rtc_unittests.
    • c221db6 : Test names changed from e.g) testOneWayVideo/chrome=chrome to testOneWayVideo/chrome-chrome.
    • 264e66f : Add encoded_timestamp to AudioEncoder base class
    • 9ea6f8a : New interface class AudioEncoder
    • 8efaa27 : Disable a bunch of Nat and Ice tests when running under DrMemory.
    • 458c2c3 : Improve rtcbot to load all test files at start and allow them to registerTests via: registerBotTest. After loading all tests main.js starts running the requested one on the command arguments.
    • 9aed002 : Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame. Controlled by setting enable_extended_processing_usage. Enabled by default.
    • d1ba6d9 : Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
    • 3e2f8ff : Selecting bot_type changed to be specified in the test file
    • e93cbd1 : Fix data races in ThreadTest.ThreeThreadsInvoke.
    • f87c0af : audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with
    • f02ba9b : audio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with
    • 8dc00d7 : audio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with
    • 99e561f : Extend AcmSwitchingOutputFrequencyOldApi with more frequencies
    • 64f5611 : Roll chromium_revision 2d714fa..de13cf4 (298667:299488)
    • fab5439 : common_audio: Removed version API from signal_processing
    • 81ddc78 : (Auto)update libjingle 77701902- 77709729
    • 1ecbe45 : (Auto)update libjingle 77689511- 77696841
    • 43336b6 : Remove unused (no-op) VideoOptions.
    • a4351a0 : libjingle: use _stricmp instead of deprecated stricmp.
    • a73a678 : Remove -1 from Call::Config::start_bitrate_bps.
    • eb24b04 : Add periodic logging of received RTP headers and estimated clock offsets for e2e delay.
    • 81a7893 : New ACM test to trigger audio glitch when switching output sample rate
    • c216b9a : Add a packet loss full stack test to the new API.
    • a57678a : Workarounds for a bug in VS2013.3 linker when PGO is turned on.
    • 7fe1e03 : Wire up external encoders.
    • f68cc0b : (Auto)update libjingle 77554188- 77629208
    • 82e6fa5 : Move exlusion of VP9 integration tests for DrMemory from modules_unittests to modules_tests file.
    • b6af428 : Adjust speech probability in NS when echo
    • 1e6a5dd : Removes xmllite from talk/xmllite since webrtc/xmllite is used instead.
    • 8bee130 : Disable VP9 integration tests on DrMemory.
    • bc1a457 : common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16
    • a3722b6 : iSAC tests: Type buffers as uint8_t[] to avoid casts
    • d4fe824 : audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with
    • 396a5e0 : WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]
    • 3f7f899 : WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16
    • 1172988 : Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]
    • 3c16d8b : (Auto)update libjingle 77414393- 77554188
    • 651c05e : Release _inputSendPin & _outputCapturePin before _captureFilter & _sinkFilter since they should depend on the filters. The previous steps work fine for all the webcam, but have problem on SplitCam driver as in the issue report. Anyway it's always good to de-initial with the reversing order to initialization.
    • 7f7b0a1 : Re-enable ThreadCheckerDeathTest.MethodNotAllowedOnDifferentThreadInDebug (missed when enabling other base tests).
    • 4ddbbed : Disable SendsAndReceivesVP9 test for now.
    • c87b747 : Adjust/increase rate control thresold for a vp9 test.
    • 573c78e : Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. Passes trybots.
    • 3cefbc9 : Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. This also marks all virtual overrides of other classes in the same files.
    • afede83 : Cleanup scripts and suppressions for TSan v1
    • fae6bc4 : Remove talk_base from suppressions.
    • e46bc77 : Reland 28629004: adding new AEC dump start interface for chrome.
    • c5593ef : Workaround deps2git issue with inline Python in DEPS.
    • c732a3e : Re-enable allmost all base tests.
    • 4a73519 : Re-enables a bunch of base unittests part II.
    • dae40dc : Change setting VP8 codec specific info values by HW VP8 encoder to follow SW implementation.
    • e30dab7 : base/thread_unittest: wrap test was setting current thread to NULL.
    • 17f8ddd : Make pbos and kjellander only owners of tsan2 suppressions.
    • 8768f16 : Fix comments in common_types.h
    • 3ff788c : Increase timeout for AsyncWriteTest.TestWrite.
    • 4bd2db9 : Opus wrapper: Use const for inputs and uint8[] for byte streams
    • 1bada48 : Make DEPS find check_root_dir.py in legacy checkouts.
    • 2c0cdbc : Estimating NTP time with a given RTT.
    • c803907 : Removing useless packets when inserting them (NetEq)
    • 0b0ac82 : Remove root_dir variable from DEPS + enforce rename.
    • 3ea35fd : common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16
    • 127ca3f : Disable TestDTLSConnectWithSmallMtu on all platforms.
    • 0001adc : Use openmax_dl on all ARM (v7 or higher) platforms.
    • 95bacfe : Remove bad waiting code from video decoder release function.
    • 97abeee : (Auto)update libjingle 77263371- 77296420
    • 536eb98 : Re-enables a bunch of base unittests.
    • 9ea5396 : Roll chromium_revision fc668e2..2d714fa (298195:298667)
    • 4165f7a : Add a variable for deciding when to use openmax_dl.
    • f71785c : audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with
    • 575d126 : Protect send_/recv_streams_ in WebRtcVideoEngine2.
    • 9c6dc46 : CHECK/DCHECK: Explicitly state whether the condition can have side effects
    • 5e3d7c7 : Change name of a NetEq internal member variable
    • 742922b : Make the media content send only if offerToReceive is false while local streams exist. We previously do not add the media content if offerToReceive is false.
    • d6bda09 : Initialize sctp_paddrparams in OpenSctpSocket().
    • 27e5898 : Explicitly unpoison FDs for MSan.
    • 46ffc70 : Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder.
    • 963b979 : Remove potential deadlock in WebRtcVideoEngine2.
    • a9e363e : Roll chromium_revision c264a05..fc668e2 (297113:298195)
    • 77d5a57 : Revert "Only configure the SSL library in one place."
    • 6ed1cf4 : Isolate: Remove use of --ignore_broken_items
    • 9103953 : Fix neteq_rtpplay so that empty SSRC is valid
    • 7cbc4f9 : Set NetEq playout mode through the Config struct
    • 8b65d51 : Add an SSRC filter to neteq_rtpplay
    • 532ed43 : Prevent reading outside iSAC bitstream, if the stream is corrupted.
    • 8234fa6 : Only configure the SSL library in one place.
    • 2fe5893 : Mac: adds missing _DEBUG flag to mac debug builds.
    • 528fc65 : Fixing build issue with L-sdk
    • 9a742b4 : talk: removes empty directories base and sound.
    • 5d3e7ac : Check on the existence of report directory
    • 42684be : Wire up CPU adaptation in WebRtcVideoEngine2.
    • 31b75ea : Moves xmllite's unittests to rtc_unittest.
    • 25cc745 : Switch to SW video decoder on Android after getting 2 or more critical errors from HW decoder.
    • 4b133da : Let RtpFileSource use RtpFileReader
    • 348eac6 : audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with
    • 5fa8c45 : Remove mouse cursor capturer from the ScreenCapturer interface
    • 6138f0f : Revert "Remove mouse cursor capturer from the ScreenCapturer interface"
    • 1fced0f : Remove mouse cursor capturer from the ScreenCapturer interface
    • 76819d3 : Add error trap for XFixesGetCursorImage()
    • 325cff0 : Import LappedTransform and friends.
    • 593c3a0 : rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
    • 4530b2c : Revert 7355 "Fix parallelization in libjingle_p2p_unittest."
    • 36b0c1a : Adds PRESUBMIT.py dispensation for depending on rtc_base.
    • fd29205 : Fix parallelization in libjingle_p2p_unittest.
    • c86e45d : Fix parallelizability in modules_tests.
    • 4cebd84 : Reland "Remove DTMF status methods from Voice Engine" r7276
    • 4e4fe4f : Add support for MSan
    • afefed5 : Update checkdeps.py rules in DEPS
    • 83fe69d : Added presubmit protecting against inclusion of rtc_base, while allowing rtc_base_approved.
    • 3037bc3 : GN: Add common configs to tools and test.
    • b8caf6a : GN: Enable libvpx, add link target and convert some test targets
    • d05756f : Changed mips_arch_variant variable value corresponding to Chromium code changes.
    • 79a7148 : Revert 7337 "Reland 28629004: adding new AEC dump start interfac..."
    • 7aad5e5 : Revert 7338 "Fixed the android build by making the interface pur..."
    • d0bb586 : Collecting stats every fixed time in webrtc_video_streaming.js test and prepare the format these collected stats to be plotted using one of external dev-tools.
    • db75a66 : Minor code change to fix some warnings in MIPS build.
    • 90d1979 : Fixed the android build by making the interface pure virtual.
    • 14092e0 : Reland 28629004: adding new AEC dump start interface for chrome
    • 792d1a0 : Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest.
    • 8752061 : Revert 7334 "adding new AEC dump start interface for chrome."
    • 2e417d6 : adding new AEC dump start interface for chrome.
    • 38c121c : Minor modifications to test::RtpFileReader
    • 1795c35 : Add default implementation of Add/RemoveObserver.
    • 65e56db : audio_processing/aecm: Added help function for calculating log of energy
    • 23ec837 : audio_processing: Removed usage of macro WEBRTC_SPL_MUL
    • 750423c : audio_processing: Replaced trivial macro WEBRTC_SPL_LSHIFT_W32 with
    • 8cad943 : Revert 7327 "Update isolate.gypi files + link to isolate_driver.py"
    • 02cd306 : Update isolate.gypi files + link to isolate_driver.py
    • 359d720 : Allow Android apps to set video renderer scaling type. Also add type check for EGL context object received from apps and switch to byte buffer video decoding if EGL context is incorrect
    • 7dfb7fa : Reland disallowing blocking calls on the worker thread. This fixed the issue that invoking the call when the thread is not started.
    • ea6c12e : Set thread scheduling parameters inside the new thread.
    • 6266240 : Disable flaky tests: JsepPeerConnectionP2PTestClient.ReceivedBweStatsCombined JsepPeerConnectionP2PTestClient.ReceivedBweStatsNotCombined
    • e794c36 : Fix parallel test execution for tools, testsupport and metrics tests.
    • d711181 : audio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16 with
    • 7c15510 : common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32
    • 24f62e1 : Adding getStats function to the exposed PeerConnection in RtcBot
    • 730d270 : Remove callback from RtpDepacketizer::Parse().
    • f21ea91 : GN: Add common configs to all targets.
    • 34f2a9e : Initialize SSL in unittest_main.cc.
    • 3a10d2f : Roll chromium_revision deaf2f7e..c264a056 (295079:297113)
    • 6c6680a : Cleanup .gclient.bot_entries to avoid sync problems on bots.
    • 3902054 : Roll chromium_revision 6455c69..deaf2f7 (293954:295079)
    • bebc75e : Fix the duplicated candidate problem when using multiple STUN servers.
    • 0a256ac : Getting orientation is not working properly. VideoCaptureImpl::RotationFromDegrees returns -1 in case fails not 0. So we need to change the if statement.
    • 5d0071f : Build one of NSS or BoringSSL but not both.
    • a21d071 : Reverting part of https://webrtc-codereview.appspot.com/15089004/diff/140001/talk/session/media/channelmanager.cc?context=10&column_width=80 because of a major regression hanging the executable on start.
    • 1fd362c : Do not assert for blocking call allowed in Thread::Join. We do not allow blocking call from the worker thread, but on Android the worker thread may stop/join a SignalThread, which hits the assert. AssertBlockingIsAllowedOnCurrentThread is used to make sure a thread does not do Invoke, so check that in Thread::Join does not seem to add much value.
    • 384d05f : Remove the different block lengths in ns_core
    • 5088377 : Revert 7297 "Remove the different block lengths in ns_core"
    • ca110b8 : Mark virtual overrides of ViENetwork and VoENetwork as such.
    • 8b2e50c : Revert 7302 "Roll chromium revision: 6455c69:2687a76"
    • bfacaab : Add accessors for array of channel pointers in AudioBuffer. They are needed as arguments to any multichannel audio processing unit.
    • b38959e : Roll chromium revision: 6455c69:2687a76
    • f1d751c : Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.
    • 0530511 : Explicitly initialize SSL for tests.
    • 61e811f : Bump to version 39
    • 60fbd65 : Removing error triggered for disabling FEC on non-opus
    • 5f39657 : Remove the different block lengths in ns_core
    • 741711a : Revert r7049/r7123, which added unnecessary "u"s to "return 0"s.
    • 3156699 : Fix typo from RtpPacketizerH264.
    • 37e1846 : Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293).
    • fe1eafb : Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.
    • 30be827 : Enable render downmixing to mono in AudioProcessing.
    • e1bba60 : Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac
    • 3987b6d : Fix a problem in Thread::Send. Previously if thread A-Send is called on thread B, B-ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A-Send to return. This caused mutliple problems like issue 3559, 3579. The fix is to limit B-ReceiveSends to only process requests from A. Also disallow the worker thread invoking other threads.
    • a0ce9fa : Call NS AnalyzeCaptureAudio before AEC
    • 70e2d11 : Reduce jitter delay for low fps streams. Enabled by finch flag.
    • 275dac2 : Moved the filter calculation from analyze to process in ns_core
    • 634c926 : audioproc: Now also writes to output file in simulation mode
    • 7ee24a7 : WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
    • d60d79a : Thread annotation of rtc::CriticalSection.
    • 38344ed : Move thread_annotations.h to webrtc/base/.
    • 8166fae : Change Android video renderer to maintain video aspect ratio when displaying camera or decoded video frames.
    • 90668b1 : Switch HW video decoder to output byte buffers if video renderer EGL context is not provided by app.
    • 1b7dcc1 : (Auto)update libjingle 76169599- 76176062
    • 94ff92c : Use VPX_IMG_FMT_*/VPX_PLANE_* defines
    • 2c1bcea : Enable ipv6 by default for webrtc under a Finch experiment.
    • 3987f10 : Revert "Remove DTMF status methods from Voice Engine" r7276
    • bf7b9e0 : Remove DTMF status methods from Voice Engine
    • e34a2e7 : Revert "Set minimum SDK level to 10.7 for Mac and iOS" (r7175)
    • faf2410 : gn: Hide modules/video_capture:video_capture_internal_impl behind an arg
    • 0e6e4d2 : Reland "Converting five tests to use new AudioCoding interface" (r7258)
    • 4f6f22f : Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface"
    • ea29787 : audio_processing/agc: Solved building with AGC_DEBUG + few style changes
    • 0a2087a : Skeleton for registering external encoders/decoders.
    • c569a49 : Unit tests for SSLAdapter
    • dc0b37d : modules_unittests: Turned on ApmTest.Process test for Android
    • a3c4d4d : Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."
    • 8c5740b : WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
    • 83f95ba : Remove engine-level SetOptions.
    • 99e404c : Revert "Converting five tests to use new AudioCoding interface" (rev 7258).
    • 35850ff : Adding test file path as argument of the rtcBot run command's arguments.
    • 64a2f10 : Remove Get/SetNetEQPlayoutMode APIs
    • 07ca949 : Adding webrtc_video_streaming test This test is streaming video and audio between two bots using webrtc js api.
    • c570761 : Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258).
    • cfe0735 : Convert AcmReceiverTest to new AudioCoding interface
    • eb1de5c : Converting five tests to use new AudioCoding interface
    • bdfdc96 : Clang-format ns_core
    • 759982d : Set number of temporal layers for VideoSendStream.
    • 6121715 : Ensure that NetEq recovers after a large timestamp jump
    • 8877287 : Disabled several rtc_unittests so the tests can be turned on in the waterfall
    • 97ed393 : Reapply 23529005 after fixing the build break issue (Chromium:582133002)
    • ed5ca1f : (Auto)update libjingle 75925673- 75926712
    • c98f217 : (Auto)update libjingle 75924589- 75925673
    • 0c9fe72 : (Auto)update libjingle 75922684- 75924589
    • ebf2757 : Fix HW video decoder crash on some Android KK devices.
    • c1eebfa : Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc.
    • e658124 : Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD.
    • fbf3bfe : Separate between Analyze and Process in NS
    • 9570560 : Additional disabled tests in rtc_unittests.
    • 34ac776 : Additional disabled tests in rtc_unittests.
    • fded02c : base: disabled several base tests on Mac so that rtc_unittests can be turned back on
    • bbe0a85 : Config struct for VideoEncoder.
    • 0268611 : Re-enable missing android tests disabled due to issue 3770.
    • 2036a7b : Clean directx_sdk_path as it is already defined in base/common.gypi
    • 5ca6008 : Creating a test helper class TimestampJumpRtpGenerator
    • 6e5c784 : (Auto)update libjingle 75875619- 75878731
    • b5a5c44 : (Auto)update libjingle 75865376- 75875619
    • d7acf11 : (Auto)update libjingle 75854833- 75865376
    • ccb3e3f : (Auto)update libjingle 75854418- 75854833
    • dcc1f04 : (Auto)update libjingle 75852725- 75853560
    • 0b435ba : A few fixes to avoid crash in HW codec on device orientation change.
    • 143ffa4 : Update iOS video capture to use non-deprecated APIs.
    • 83af77b : Revert maximum video codec resolution on Android back to 720p again.
    • 933d88a : (Auto)update libjingle 75818332- 75837294
    • c3091a6 : Remove the 'webrtc_test_video_render_dependencies' target.
    • 42731bd : Avoid writing a double/float to a string to avoid a crash.
    • ba737cb : Do not require synchronization access on the thread if called from rtc::Thread::WrapCurrent. The synchronization access is unnecessary for rtc::Thread::WrapCurrent (called from JingleThreadWrapper) since JingleThreadWrapper never calls rtc::Thread::Stop or rtc::Thread::Join. Failing to get the access caused crashes in Chrome since rtc::Thread::Current will be NULL when rtc::Thread::WrapCurrent fails.
    • 6116062 : Trying to fix Chrome FYI bots.
    • e94f83a : Cleanup .gclient_entries to avoid sync problems.
    • 205c15a : Adds asan suppresions for rtc_unittests
    • 6cd6ba8 : Expose VP8/H264 defaults through video_encoder.h.
    • c7134f8 : Fix proper deps in BUILD.gn files. This should make Chrome GN bots happy.
    • fda2c2e : Add Analyze API to NS
    • ab071da : Split video_render_module implementation into default and internal implementation. Targets must now link with implementation of their choice instead of at "gyp"-time.
    • 369a637 : Implemented Network::GetBestIP() selection logic as following.
    • 3b67f8e : Enable HW video decoding on Qualcomm devices.
    • d91608d : The 2x2 black frame on windows when the shared window is minimized caused an assert from vp8 and may lead to memroy corruption. It's changed to 1x1 to match the behavior on Mac. The Chromium code will detect the size and convert it to a black frame in the original size.
    • 5422e72 : Modifying NetEqExternalDecoderTest
    • 4a5061f : talk/p2p/base: removed unused variable "port_"
    • 5a098c5 : Refactor VP8 de-packetizer.
    • 3bd5603 : Revert "Disable video_capture_tests for Android." (revision 7023).
    • a74eda1 : Split video_capture_module specific implementation (external vs internal capture) into its own targets. Dependencies must link directly with the desired one.
    • 85ef770 : Split video engine android initialization into each internal module initialization.
    • ab990ae : Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
    • 6a9b155 : (Auto)update libjingle 75683337- 75695882
    • e387cc0 : webrtc/overrides: add OWNERS-file.
    • dc8dcb4 : Narrower include for constructormagic.h in Chromium.
    • eb43264 : Remove linux_memcheck from default trybots.
    • a59c501 : Java VideoRenderer class may be backed by two different native classes depending on type of rendering. Fix crash in AppRtcDemo by calling correct destructor on exit.
    • 40c2aa3 : Implemented Network::GetBestIP() selection logic as following.
    • f8bff76 : Implemented Network::GetBestIP() selection logic as following.
    • 7351d4d : Add a gyp target for producing a voice engine merged library.
    • a6cefca : gn: Fix cflags usage
    • cddd17c : Recreate VideoStreams when setting resolution.
    • 88e85ad : Add pbos@webrtc.org (myself) to talk/media/webrtc/.
    • dae612e : Mark all virtual overrides in the hierarchies of UdpTransportData and UdpSocketWrapper as such.
    • 80132e4 : (Auto)update libjingle 75610402- 75610402
    • 699c46a : rtc_unittest: prevent execution of broken tests.
    • 4436020 : Fix GN for rtc_base_approved target.
    • 178015d : memcheck: suppressions didn't map over directly when moving base from talk to webrtc (part of the suppression that is not related to the signature differed). Fixed suppressions accordingly.
    • 595b23c : Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..."
    • c75f607 : audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2
    • 6ae5a6d : Add a target for the approved subset of rtc_base.
    • b3cbeb3 : Fix memory leak in webrtc::MouseCursorMonitorMac
    • ab7073a : Partial implementation of rtc::LogMessage in chromium overrides.
    • 9967845 : HW video decoding optimization to better support HD resolution:
    • cd309e3 : Enable ipv6 by default for webrtc under a Finch experiment.
    • 000d867 : Make BW checks 0 in peerconnection_unittest.cc.
    • 7bb2586 : audio_processing: Correct sample rate in aec_debug_dump
    • 76ba7ca : Re-enable neteq_performance_unittest.cc for android.
    • 541753f : Re-enable rampup_tests.cc for Android.
    • 4a6c5b3 : Re-enable video send stream tests for android.
    • 18617cf : Fix ThreadChecker unittests when DCHECK_ALWAYS_ON is defined
    • 7f82635 : Stop building talk/xmllite since it is no longer used.
    • 192ab71 : Set minimum SDK level to 10.7 for Mac and iOS.
    • a42a3ad : (Auto)update libjingle 75390072- 75428737
    • 7e31197 : Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..." BUG=3789 TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
    • 91ee746 : Add enable flag for Android device orientation change event.
    • 192a54f : Temporary revert maximum video codec resolution back to 1080p.
    • 3decd9b : Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
    • 1fb5d12 : Initialize restored_packet in nack_rtx_unittest.cc.
    • c3c9015 : linux: remove stray libcrypto dependency
    • 78b2d56 : Disable MethodNotAllowedOnDifferentThreadInDebug.
    • d2cf48d : Fix mac video_render implementation on cocoa.
    • f7e5f22 : Fix stack limit exceeded in http client.
    • a0d7827 : Add ability to downscale content to improve quality.
    • b5e6bfc : Make RTPSender/RTPReceiver generic.
    • 6071b06 : Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.
    • cc774a6 : Mark all virtual overrides in the hierarchies of RtpDump and VCMPacketizationCallback as such.
    • ea77334 : (Auto)update libjingle 75302540- 75327856
    • 31c285b : Update AUTHORS file.
    • 8995996 : Fix window capturing on Windows when the window is minimized.
    • f520ea5 : Skip dlclose() on AddressSanitizer.
    • 1d8f780 : Stop building talk/sound since it is no longer used.
    • 1d53f64 : Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD.
    • b990674 : Split suppressons of thread.cc and messagequeue.cc.
    • 4b049fc : Remove developing code in ns_core
    • f5bdd54 : Add myself to common_audio and audio_processing watchlists
    • 307d3db : Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
    • 665d861 : Restore webrtc_base target until r7140 is rolled into Chromium.
    • 8dd60cc : audio_processing_unittests: Enabled ApmTest.Process for all platforms but Android
    • c665dcb : Revert 7145 "Stop building talk/sound since it is no longer used."
    • 2b58a44 : Calculating round-trip-time in send-only channel in VoE.
    • 1972ff8 : Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
    • 4c87645 : Stop building talk/sound since it is no longer used.
    • 47658f1 : Mark all virtual overrides in the hierarchy of AudioPacketizationCallback, RTPStream, and NetEq as such. Also mark all other virtual overrides in the same files.
    • 1711104 : Fix MSVC warnings about value truncations, webrtc/base/ edition.
    • 3472dcd : Fix frame rate selection for Android camera.
    • 67eabc0 : Add schannel webrtc_base build using a new use_schannel gyp variable.
    • b2efb67 : Put base tests in webrtc_tests.gyp
    • a8d2ee7 : Roll chromium_revision ea769fd..6455c69 (re-land)
    • b6d6928 : Enable shared socket for TurnPort. In AllocationSequence::OnReadPacket, we now hand the packet to both the TurnPort and StunPort if the remote address matches the server address.
    • 0867f69 : Convert GN visibility to be lists.
    • 5c20bb2 : Remove suppressions for VideoFrame::Validate.
    • 33aa095 : Simplify gyp rules on video_render_module.
    • e0761d0 : Fix printing of error stack in rtcbot when a test fails via test.fail().
    • 49fa212 : Fix compile error on JDK 1.7.
    • 0fa0475 : Roll gtest-parallel.
    • 23a5e3c : Remove DestructEncoderInst and its codec-specific implementations.
    • a2e6a52 : Revert 7128 "Roll chromium_revision ea769fd..6455c69"
    • 5d639b3 : (Auto)update libjingle 75141932- 75179475
    • fdba9ee : Roll chromium_revision ea769fd..6455c69
    • 4ca66d6 : include cstdlib for free() and abort()
    • fa60398 : Add a new class InterfaceAddress inherited from IPAddress to keep track of IPv6 Address flags.
    • 87ff9c8 : Fix up configs applying to GN build.
    • 7d4891d : Fixes two issues in how we handle OfferToReceiveX for CreateOffer: 1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent. Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
    • a941970 : Change explicit static cast from int to uint16_t to implicit cast of 0u. BUG=3663 TESTED=local windows build with VS2013. R=harryjin@google.com, tina.legrand@webrtc.org
    • 9fe1101 : Fix the RTC+Chromium GN build.
    • 54cf150 : ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that. BUG=3789 TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate* R=tommi@webrtc.org
    • 22406fc : TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
    • 04b853b : Bot Browser files moved to /bot/browser/
    • 3d81b1b : Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got reverted due to some internal compile failures.
    • 4bbd3c8 : fix a bug in the logic when new Networks are merged. This happens when we have 2 networks with the same key
    • 1b088ee : More suppressions, uninitialized read in cricket::VideoFrame::Validate
    • 4d19e05 : Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own.
    • b420191 : Expose VideoEncoders with webrtc/video_encoder.h.
    • 641bda6 : Initialize ChannelBuffer's memory to avoid uninitialized reads.
    • 8b0b211 : Revert 7093: "Implementing ICE Transports type handling in libjingle transport."
    • 519c9e2 : Convert GN visibility to be a list.
    • 7118e61 : Finish work queue in SctpDataMediaChannelTest.
    • 0e52772 : Fix a bot-breaking memory leak from early returning in ParseMediaDescription.
    • c172320 : Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android.
    • 17454f7 : Add ctors to ChannelBuffer to enable copying on construction.
    • fd42f9d : (Auto)update libjingle 74955991- 75042522
    • 1272ee5 : Suppress uninitialized read warning in cricket::VideoFrame::Validate
    • c64246f : Set a default speech type in iSAC wrapper
    • ed8bcd3 : Starting to implement the new ACM API
    • 9600519 : Adding the ability to test on Chrome for Android. use "android-chrome" as type in rtcbot running command. Example: node test.js android-chrome
    • 37c39f3 : audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16
    • 0d394f3 : video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16
    • c77e4d6 : - Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices.
    • 142bb9d : Roll chromium_revision 94532b1..ea769fd
    • fe16167 : Fix RTT calculations for send-only channels.
    • c30e9e2 : Ignore FEC packet in stats, if it is first packet on ssrc.
    • 6d08ca6 : GN: Prefix WebRTC specific variables with "rtc_"
    • f68cf93 : Add video_capture_tests_apk_target
    • 7256d31 : Implementing ICE Transports type handling in libjingle transport.
    • a781f68 : Fix rm command for class cleanup in r7091
    • 9510022 : Cleanup temporary class files for OpenSlDemo
    • cc06056 : Remove unnecessary include from testutils.cc.
    • 992febb : (Auto)update libjingle 74873066- 74873164
    • a3344cf : Fix webrtcvideoframe tests.
    • ddb85ab : Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07
    • 8f073c5 : Create a new interface for AudioCodingModule
    • af5fa95 : (Auto)update libjingle 74857067- 74860820
    • 7e3bd3d : (Auto)update libjingle 74851128- 74857067
    • bc6fa18 : (Auto)update libjingle 74825992- 74851128
    • 287e961 : Disable TestDrain test on memcheck bots.
    • cdb48db : Enable VideoAdapterTest.BlackOutput on DrMemory.
    • fed47dc : Drop buildbot_tests.py script
    • a2da031 : Remove use_relative_paths from DEPS
    • bcf75e3 : Modifying audio_coding/codecs/OWNERS
    • c2c4117 : common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with in audio_processing
    • 2c03a97 : Roll chromium_revision f0a439d..94532b1
    • 818b7b3 : (Auto)update libjingle 74825084- 74825992
    • dfbcf81 : Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice.
    • f1427c6 : Revert 7070 "TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH."
    • 4b23404 : Reduce maximum video resolution for Android.
    • 574f2f6 : TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
    • 021e76f : Add support for WAV output in audioproc
    • 52055a2 : Fixes two issues in how we handle OfferToReceiveX for CreateOffer: 1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent. Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
    • afa77cd : Add direct_dependent_config to desktop_capture in GN build.
    • ceb956b : Abort Negotiate() if DoCreateOffer() fails.
    • d57c95f : Change Chromium .gclient to not use Managed mode.
    • fa822b9 : Fix strange owners files with comments that crashs "git cl presubmit"
    • 79ee97c : [MIPS] Fix gn gen failure for MIPS in webrtc
    • 38ef664 : Moving the api.js and bot.js to /rtcbot/bot/ to be shared between /borwser and /android
    • 262e676 : Reland rev 7041 with BUILD.gn files.
    • 3cbd6c2 : Fix MSVC warnings about value truncations, webrtc/common_audio/ edition.
    • f6ab6f8 : Rename Audio[Multi]Vector.CopyFrom to .CopyTo
    • 3c0aae1 : Change gflags and gmock includes to be full paths.
    • 51bb33c : ACMOpus: Remove useless member variable fec_enabled_
    • 7825b1a : Add support for multi-channel DTMF tone generation
    • bcb6bcf : Remove HybridVideoEngine.
    • 9d45393 : Change return value for number of discarded packets to be int.
    • 01581da : Fix audio/video sync when FEC is enabled.
    • bfd7a8c : Fix compile errors on webrtc/base.
    • 0229cba : Remove ambiguous call to MakeCheckOpString.
    • 95c2458 : * Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files.
    • 9328f39 : cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error BUG=3663 TESTED=ninja local build on windows. R=andrew@webrtc.org, kwiberg@webrtc.org, thorcarpenter@google.com
    • 5b83af4 : Fix leak of NSAutoreleasePool.
    • 609f987 : (Auto)update libjingle 74696326- 74723281
    • 1b8b4c4 : Revert 7041 " Audio codecs to include webrtc/typedefs.h"
    • fa4535b : (Auto)update libjingle 74694022- 74696326
    • 26c0c41 : Network up/down signaling in Call.
    • ebee401 : Remove flake in SendsLowerResolutionOnSmallerFrames.
    • c4175b9 : Set resolution based on incoming VideoFrames.
    • 9730d3a : Audio codecs to include webrtc/typedefs.h
    • 0372b93 : Partial revert of r7014 (Android APK refactor)
    • bac0726 : Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes
    • adee8f9 : Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate
    • 0a214ff : Setting marker bit on DTMF correctly
    • 74cf916 : Fix issues in audioproc for float aecdumps
    • 48f2568 : audio_processing/nsx: Bug fix that could cause divide by zero
    • d944a68 : Suppressing VideoAdapterTest.AdaptResolutionWide and VideoAdapterTest.AdaptResolutionNarrow on DrMemory
    • 72e4485 : (Auto)update libjingle 74628537- 74648573
    • 9075048 : Remove deprecated RTCVideoRenderer constructor.
    • 34a6764 : Remove the checks.h dependence on logging.h in a standalone build.
    • 8e24d87 : Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator from both the audio playback thread and the network thread without locking.
    • 9f34128 : Remove WebRtcVideoEngine::default_codec_format().
    • 0365514 : Remove files from talk/PRESUBMIT.py.
    • d72a759 : Create a copy of talk/xmllite under webrtc/xmllite.
    • 6f729e8 : Disable video_engine_tests and webrtc_perf_tests on Android.
    • ee0fb18 : Divide-by-zero problem in NetEq's Normal::Process fixed
    • 94da203 : Remove retired android_apk[_rel] trybots from PRESUBMIT.py
    • 324b72d : Disable video_capture_tests for Android.
    • e281f7f : GN: Update webrtc/base to recent GYP changes.
    • 468516c : RTCBot is a framework that allows to write tests where logic runs on a single host that controls multiple endpoints ("bots"). Thus allowing to create more complex scenarios that would otherwise require non-trival signalling between multiple parties.
    • 561a9ec : Update checkedeps.py rules in DEPS.
    • 76a4257 : Remove build_with_chromium==1 conditions for Android
    • 841f58f : Unpacking aecdumps generates wav files
    • c3f42f3 : Fix audio_decoder_unittests.isolate
    • 8dbeb5b : Adding more codecs to the AcmSenderBitExactness
    • 7e86049 : Roll chromium_revision 681cc8e..f0a439d (r292217:r292861)
    • 3bd4156 : Android APK tests built from a normal WebRTC checkout.
    • c4870bb : GN: Audio device module
    • 524b8f7 : GN: Implement voice engine, common audio, audio coding and audio processing
    • 1b9a188 : GN: Fix webrtc/video/BUILD.gn for Chromium build.
    • a22485e : MIPS optimizations for AEC audio processing module
    • af7fdfc : Add LTO support for Android Chromium.
    • f554d75 : Allow same src and dst in InputAudioFile::DuplicateInterleaved
    • 44010f3 : win: Replace custom assert() macro with regular assert.h
    • bc3f333 : Add jiayl to talk OWNERS.
    • e21cc9a : When the peerconnection creates the offer with a constraint to disable the audio offering, stats will not get properly updated.
    • b0dc3d7 : Precompile out our standalone CHECK macros in a Chromium build.
    • a5b7869 : Add CHECK and friends from Chromium.
    • 11c6bde : Specify an ECDH group for ECDHE.
    • 55e9da1 : Add talk owners to migrated talk folders
    • 4431fd6 : Add 60 fps video support
    • 788f058 : GN: Implement video_engine, video_capture and video_render.
    • df9fef6 : common_audio: Removed macro WEBRTC_SPL_DIV
    • 1f8a237 : (Auto)update libjingle 74235596- 74297316
    • 59a1b1b : Fix the different samples per channel in aecdump
    • deaece6 : Disable VideoAdapterTest.BlackOutput on DrMemory.
    • f8723d6 : Add unit tests to rtcp_receiver_test.
    • 2dbb47a : Roll chromium_revision b1748b:681cc8
    • 956f281 : Re-enable all VideoAdapterTests on DrMemory.
    • 75c3ec1 : Fix data races during VideoAdapterTest tear-down.
    • 573a1ee : (Auto)update libjingle 74202294- 74230205
    • 18584fc : Move end of namespace inside #ifdef
    • c3c2911 : Expose setPayloadType on the rtp_sender. Thus allowing other users of this module to set the payload type to be used without having to call SendOutgoingData.
    • 00f11f5 : - Make local constant non-static. - Remove spammy log line.
    • 66a3582 : Create a copy of talk/sound under webrtc/sound.
    • 7087857 : implement handling ALTERNATE-SERVER response from turn protocol as specified in RFC 5766, also created 2 test cases for both the normal redirection case as well as when a pingpong situation happens, the allocation should fail
    • dc926a0 : Avoid syncing unnecessary Chromium deps for WebRTC.
    • 3533bfc : (Auto)update libjingle 74132319- 74133664
    • 4470d78 : (Auto)update libjingle 74128148- 74132319
    • b623c5c : Disable EndToEndTest.RestartingSendStreamPreservesRtpState in video_engine_tests because it is flaky
    • f21ac1f : Fix Win64 compile of videoadapter_unittest.cc.
    • c9b3f77 : Fix data races in VideoAdapterTest.
    • 8940ce7 : Updating svn:ignore entries
    • b648b9d : Remove test constructor in WebRtcVideoEngine2.
    • 4f71e22 : Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV
    • 1de0cc4 : common_audio: Re-enable WebRtcSpl_AddSatW32() and WebRtcSpl_SubSatW32() optimizations on armv7
    • 047a46f : Remove Android.mk build files.
    • b96ea2a : Remove former team members from OWNERS and WATCHLISTS
    • 204cd56 : (Auto)update libjingle 74064646- 74072040
    • e9bfed0 : Move constant so it is not stripped out for TSAN bots.
    • 857130f : (Auto)update libjingle 74039473- 74044292
    • 79ad37e : Update root OWNERS file
    • 6556a59 : As expected, r6569 (https://code.google.com/p/webrtc/source/detail?r=6965) caused memcheck bots to complain. Adding expections for that, in line with outher peerconnection tests.
    • c239234 : Roll chromium_revision 289723:291647
    • 42ee5b5 : GN: Disable Chromium clang plugins for standalone build.
    • b4c7b09 : (Auto)update libjingle 73927775- 74032598
    • 926707b : Refactoring common_audio: Replace trivial multiplication macro
    • d32c438 : Re-landing r6961
    • 4a616be : Revert 6961 "common_audio/signal_processing: Remove macro WEBRTC..."
    • 4f01017 : common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8
    • 6e71d17 : Refactoring common_audio/signal_processing: Replaces trivial macros
    • 584cd8d : Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16-float conversion)
    • 3740d74 : (Auto)update libjingle 73927658- 73927775
    • 309a611 : (Auto)update libjingle 73891518- 73927658
    • 2b0554f : (Auto)update libjingle 73794259- 73891518
    • 97fdeb8 : Remove static initializer in WebRtcVideoEngine2.
    • 374d39b : Increment sync_chromium.py version to force re-sync
    • 1613638 : Make the last_sync_chromium file a bit more comprehensive.
    • 153c616 : Landing issue 15189004
    • 7bd5fef : Making sure muc members get recorded.
    • 038cee2 : Add send-side bit-exactness test for AudioCoding Module
    • 9b8102c : Use a deterministic input in NetEqBgnTest
    • 6b2659c : Refactoring common_audio/signal_processing: Remove unused macro WEBRTC_SPL_MUL_32_32_RSFT32BI
    • 905f9ef : Fix clang -Wformat warnings.
    • add54ad : Convert nsx_core_neon.S to unified syntax.
    • 286210d : Use --gclientfile instead of --spec, because windows is THE WORST.
    • 98d92d6 : Make sync_chromium use the git-cache when on the bots.
    • 8dcf61f : Bump WebRTC version number. Starting now, we will be setting WebRTC major version numbers to align with Chrome.
    • 3aa837c : Increase verbosity for gclient sync of Chromium
    • bbca4dd : Pass --verbose to gclient sync of Chromium
    • 8925662 : Make WebRTC work with Chromium Git checkouts
    • 3fb2d0c : Add TSAN suppression for heap-use-after-free in libvpx
    • 58c1c98 : Remove DEPS reference to third_party/clang_format
    • 5227534 : Refactoring common_audio: Remove macro WEBRTC_SPL_MEMMOVE_W16
    • 6908b84 : Disable two tests in TurnPortTest
    • 95bbd18 : (Auto)update libjingle 73627179- 73695227
    • 877083c : New utility class for easy debug dumping to WAV files
    • 71d9572 : Minor bug fix and cosmetic changes in AEC MIPS optimizations.
    • 5a60aed : (Auto)update libjingle 73626701- 73627179
    • 84532e5 : (Auto)update libjingle 73626167- 73626701
    • 0481f15 : (Auto)update libjingle 73399579- 73626167
    • d5b292e : Active connection stats [LocalAddress,RemoteAddress,LocalCandidateType...etc] is now printed in the head-up display in Android appRTC.
    • 742bac2 : Remove __inline from WebRtcIsacfix_Log2Q8.
    • 544f647 : webrtc/base: removes accidental #error in r6909.
    • 047abc9 : Remove trailing null character from std::string
    • a1ad844 : Precompute the AEC FFT tables, rather than initializing at run-time.
    • 4a25199 : GN: Fixes for Chromium builds.
    • d798095 : replace inline assembly WebRtcNsx_PrepareSpectrumNeon by intrinsics.
    • f86b262 : MIPS optimizations for ISAC (patch #3)
    • 3ea9f57 : Remove unneeded WebKit dependency from DEPS.
    • e9b493e : Removing macro in acm_opus.cc
    • b5ab52d : common_audio/signal_processing: Remove unused macros WEBRTC_SPL_GET_BYTE and WEBRTC_SPL_SET_BYTE
    • 8a2c84f : Log the Android Audio API choice correctly.
    • d235eae : Suppress deprecation warnings in video_capture for iOS
    • 34a865a : Roll chromium_revision 288251:289723
    • d402875 : Set updated_rect for frames generated by WindowCapturer implementationsw
    • 353cd37 : (Auto)update libjingle 73370064- 73399579
    • fb1eb43 : Rename linuxwindowpicker to x11windowpicker & only use it with use_x11
    • 5b06b06 : Revert 6897 (i.e. Reland 6863) - "Revert 6863 "Refactor StatsCollector and associated..."
    • 1e3ef4b : common_audio/signal_processing: Remove macro WEBRTC_SPL_UMUL_32_16_RSFT16
    • a84b0a6 : Small refactor on ViE to remove redudant conditions and long ifdefs.
    • 2e18638 : Exlcude two tests in VideoAdapter for WinDrMemoryFull.
    • 58e2d26 : Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().
    • e8018b0 : Adding a 5% as packet loss level for Opus
    • 4521e2d : Adding online bitrate change to voe_cmd_test
    • 817a034 : Fix TimeToSendPadding return to be 0 if no padding bytes are sent.
    • 8434dbe : common_audio/signal_processing: Remove macro WEBRTC_SPL_SUB_SAT_W32
    • c3df61e : (Auto)update libjingle 73256845- 73260148
    • 22fa032 : Revert 6863 "Refactor StatsCollector and associated types."
    • 449ad98 : (Auto)update libjingle 73248599- 73249894
    • ef8bb8d : Make sure that muting muted streams succeeds.
    • 432893a : Remove TODO saying to remove WebRtcVideoFrame.
    • b15dddf : Remove files from talk/PRESUBMIT.py blacklist.
    • d968dd0 : Fixes failure triggered by include order re-ordering.
    • a09a999 : (Auto)update libjingle 73222930- 73226398
    • b242c44 : Further DrMemory suppressions, likely from r6811
    • 2c0fb05 : (Auto)update libjingle 73221069- 73222930
    • 67f8495 : (Auto)update libjingle 73215194- 73221069
    • 4eeeefe : (Auto)update libjingle 73072800 - 73215194
    • 23a4d85 : Decreased kMaxOverusesBeforeApplyRampupDelay (from 7 to 4). Increased kStandardRampUpDelayMs (30 to 40s).
    • 38d8881 : Fix the audio source failure due to unsupported constraints.
    • 5af76ae : Removing TODOs related to AcmReceiverBitExactness checksums
    • 388bd79 : Update checksums for AcmReceiverBitExactness on android
    • 023f12f : NetEq background noise generation off by default
    • c27543d : Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field.
    • e999bd0 : Removing ASSERT for tcp candidate for port 0 and 9, as Android clients may not be called with set_allow_tcp_listen(false).
    • afb554f : Move default-recv-channels to a separate class.
    • c891fee : Make a int64 constant use ULL suffix so it wont get truncated. BUG=3690 TESTED=try bots R=andrew@webrtc.org
    • c6273b5 : DrMemory suppresssions, likely from r6811.
    • c3d2bd2 : Fix GetStats() crash.
    • 3d53f61 : .gitignore removed openssl
    • aa2344e : talk/third_party: removes the empty directory.
    • 8d57f08 : (Auto)update libjingle 73072800- 73072800
    • 40995c7 : Fixing uninitialized variable in file_audio_device.cc.
    • 0a3cbb3 : common_audio/signal_processing: Removes macro WEBRTC_SPL_MUL_32_32_RSFT32
    • cf8f33a : Removes mismatching signs in signal_processing_unittests
    • 6aac93b : Adding SetOpusMaxBandwidth in VoE and ACM
    • c98ce3b : modules/audio_processing: Updates output_data_fixed.pb test file
    • 6ac22e6 : Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798
    • 820f8e9 : modules/audio_processing: Moves declaration of kDelayDiffOffsetSamples
    • 065247b : Rebase webrtc/base with r6863 version of talk/base: cls integrated: r6809 svn diff -r 6808:6809 http://webrtc.googlecode.com/svn/trunk/talk/base 6809.diff patch -p0 -i 6809.diff
    • 730bf30 : Refactor StatsCollector and associated types. * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase. * Reports are now managed in a set, not a map, since it's enough to store the id in one place. * Report ids are now const. * Copying of data has been greatly reduced. * This change includes preparation work for making GetStats fully async.
    • 1c83912 : Use test::Packet test::PacketSource classes in neteq_rtpplay
    • 96d8b0e : Revert 6860 "SSE2 version of SubbandCoherence()"
    • 0db82f3 : SSE2 version of SubbandCoherence()
    • 7ec3f9f : Fix a bug in parsing IceCandidate with IPV6 address. It used to treat ":" as a candidate delimiter and got confused by the ":" in the IPV6 address. The new logic is to check if the input has multiple lines. If so, returns error.
    • 9eabe5e : (Auto)update libjingle 72931377- 72931377
    • 2d60c5e : Encoding and Decoding of TCP candidates as defined in RFC 6544.
    • 8c01e59 : Allow root build dependencies to be overridden.
    • 53df88c : (Auto)update libjingle 72847605- 72850595
    • 65b98d1 : (Auto)update libjingle 72839629- 72847605
    • 3763b9b : webrtc/base: removes linkage of crypto
    • c8554be : Support for TURN/TLS.
    • cb46de2 : Add new OWNERS file to talk/examples.
    • 5b1ebac : (Auto)update libjingle 72820109- 72822008
    • d509678 : (Auto)update libjingle 72819313- 72820109
    • 94b996c : (Auto)update libjingle 72785516- 72819313
    • 59a2f9f : Remove the old H264 code now that a new H.264 packetizer has been implemented.
    • 9d74f7c : Fix single nalu packetization bug.
    • e8c84bf : Fix so video_replay logs aren't spammed.
    • 1d956dd : Since the packet loss rate cannot be estimated accurately, there is always a mismatch between the estimated packet loss rate and the true one. Such a mismatch will make Opus FEC suboptimal.
    • ea25784 : Change how background noise mode in NetEq is set
    • 476efa2 : (Auto)update libjingle 72785180- 72785516
    • 4f0d401 : (Auto)update libjingle 72682155- 72785180
    • aaecefe : Revert 6839 "Allow root build dependencies to be overridden."
    • e34abfb : Allow root build dependencies to be overridden.
    • 4b5625e : RTP video playback tool using Call APIs.
    • 1ccff34 : Fix crashing fake network pipe tests.
    • 2a8df7c : Fixing two bugs in voe_cmd_test.
    • 79c3359 : Add end-to-end H.264 packetization test.
    • e415864 : GN: Add PRESUBMIT.py check for GN changes + default bots.
    • 8b033ad : Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant.
    • 56d8e05 : A followup to r6828 to fix a condition check in mediasession.cc.
    • d7b4dea : initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket.cc to fix build error on vs2013 BUG=3660 TESTED=set DEPOT_TOOLS_WIN_TOOLCHAIN=0 & set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 &ninja -C out\Debug R=andrew@webrtc.org
    • dde16f1 : Fix some code styles.
    • 624a504 : (Auto)update libjingle 72659510- 72673987
    • e7d47a1 : Maintain the order of the m-lines in CreateOffer and CreateAnswer. The order in the offer follows the order in the current local description. The order in the answer follows the order in the current offer.
    • e086af0 : Fix implicite cast from signed int to unsigned int in unittest.cc BUG=3636 TESTED=set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Debug R=pthatcher@webrtc.org
    • 923db6a : Remove remove_old_gn_binaries DEPS entry.
    • fdcb42d : Fix potential crash when depacketizing VP8.
    • 8e88599 : (Auto)update libjingle 72566057- 72591796
    • d654285 : Unbreaks linux.cc in Chromium.
    • b18bf5e : Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer. Constraints are still supported for CreateOffer, but converted to RTCOfferOptions internally.
    • b01ce14 : add some comments about DEPS lkgr for chromium BUG=none TESTED=none R=harryjin@google.com
    • c9b5072 : DrMemory suppression due to r6811.
    • ee135f7 : Memcheck suppression. Re-suppress 3478 suppression after namespace change from talk_base to rtc.
    • a27342b : (Auto)update libjingle 72446860- 72550257
    • 0040a6e : This is a setup to solve https://code.google.com/p/webrtc/issues/detail?id=1906
    • 84b9e1e : Fix for retransmission. Base layer packets were not retransmitted. Issue introduced in r6669.
    • e0d03f1 : (Auto)update libjingle 72443101- 72446860
    • 6e203d5 : (Auto)update libjingle 72442050- 72443101
    • 52148c2 : (Auto)update libjingle 72430895- 72442050
    • 7cb60cc : (Auto)update libjingle 72407428- 72430895
    • 3bc4824 : (Auto)update libjingle 72403605- 72407428
    • 6955213 : (Auto)update libjingle 72389720- 72403605
    • 42d65ce : Fix memory leak in FakeSSLCertificate::GetChain(), discovered by Linux Memcheck build/try bots.
    • 1a678c6 : (Auto)update libjingle 72320533- 72380285
    • 6b21b71 : (Auto)update libjingle 72205295- 72320533
    • e1c9caf : Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804.
    • 2ec5606 : Add H.264 packetization.
    • bfe6e08 : Add simulation of network effects to video_loopback tool.
    • d9843da : libjingle: stop building files in talk/base as they are no longer used as of r6799
    • 48305f5 : Disable warning 4702 which affects map, xlist and others on vs2012 and vs2013. BUG=3584 TESTED=python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Release R=pthatcher@webrtc.org
    • 901debd : roll libyuv to r1038 from r1035 to add gyp define that makes jpeg optional. BUG=libyuv:346 TESTED=set GYP_DEFINES=target_arch=ia32 libyuv_disable_jpeg=1 R=andrew@webrtc.org
    • d4e598d : (Auto)update libjingle 72097588- 72159069
    • fc8b087 : Remove dependency on openssl for android, add dependency on boringssl. Should make Android bots green again.
    • fdbe144 : Use C functions in aec for MIPS
    • e75d78d : Integrate rtcp packet class to rtcp receiver tests.
    • 51c5508 : (Auto)update libjingle 72016417- 72097588
    • 8aed945 : Remove a disabled test.
    • 4fe98a9 : Remove clang-format rm_binaries.py DEPS entry.
    • 961293d : webrtc/base: FileModifyTime - OlderThan as that's what it was ever used as. Needed for cl/70828325.
    • af9e794 : Fix compilation on windows with clang, indentation cleanups
    • 257e130 : Set NACK/REMB when setting receive codecs.
    • 3155f2b : Roll chromium 282879:285412.
    • 185636c : Revert of 6778 "Refactor StatsCollector and associated types." Breakes FYI bots.
    • c7b8f39 : Fixes "argument list too long" problem on Linux by using the "find" command instead of re-implementing one in python.
    • 1ebd2e9 : Remove timestamp retreival warning/error.
    • 2386882 : Revert "Fix compilation on windows with clang, indentation cleanups"
    • a44fce5 : Fix compilation on windows with clang, indentation cleanups
    • 190d269 : Refactor StatsCollector and associated types. * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase. * Reports are now managed in a set, not a map, since it's enough to store the id in one place. * Report ids are now const. * Copying of data has been greatly reduced. * This change includes preparation work for making GetStats fully async.
    • 06b04ec : Fix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport.
    • f946068 : Make sure padding is sent on the first sending RTP module.
    • 45304ff : (Auto)update libjingle 71829282- 71834788
    • 39f831f : Re-revert of 6747 "Refactor StatsCollector and associated types." Breakes FYI bots.
    • 437d57d : (Auto)update libjingle 71775619- 71778545
    • 8c7e329 : Revert 6747 "Refactor StatsCollector and associated types." Breakes FYI bots.
    • 8721f98 : Revert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl."
    • e2da234 : (Auto)update libjingle 71766184- 71775619
    • 21b4da8 : (Auto)update libjingle 71753329- 71766184
    • 0f7328c : Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl.
    • 9359cb3 : Enable SendAndReceive tests.
    • f24c4a3 : Fix flaky ramp-up test.
    • 5ff71ab : Revert "(Auto)update libjingle 71675033- 71726409"
    • 89c833c : (Auto)update libjingle 71726409- 71726772
    • f67f6aa : (Auto)update libjingle 71675033- 71726409
    • 8120353 : Implement suspend-below-min-bitrate option.
    • 543e589 : Wire up VideoOptions for payload-based padding.
    • efe4b9a : Add VP8 video decoding hw acceleration support to Java Peerconnection library. For now NVidia decoder is supported only, Qualcomm will be added once b/16353967 is fixed.
    • 6f48f1b : Implement encoder options in WebRtcVideoEngine2.
    • cadd078 : Remove unused config.h and math.h includes.
    • 194fea7 : The lastest commit on this file was in
    • 85f4294 : Enable ReceiveStreamReceivingByDefault test.
    • b0c8228 : Remove no longer used SkipEncodingUnusedStreams.
    • 5ab7616 : Remove remains of WEBRTC_NO_STL.
    • fa5fcd6 : (Auto)update libjingle 71599033- 71605904
    • e69b061 : (Auto)update libjingle 71575585- 71599033
    • ceafa8c : MIPS optimizations for ISAC (patch #2)
    • 908f57e : Disable GetStatsForInvalidTrack while I rewrite it.
    • 756b846 : Refactor StatsCollector and associated types. * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase. * Reports are now managed in a set, not a map, since it's enough to store the id in one place. * Report ids are now const. * Copying of data has been greatly reduced. * This change includes preparation work for making GetStats fully async.
    • fd61a1d : Revert 6745 "Refactor StatsCollector and associated types." Broke build on android.
    • 647e05c : Refactor StatsCollector and associated types. * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase. * Reports are now managed in a set, not a map, since it's enough to store the id in one place. * Report ids are now const. * Copying of data has been greatly reduced. * This change includes preparation work for making GetStats fully async.
    • 3c10758 : Check before send/receive rtp header extensions.
    • 8fdeee6 : Implement Base::ConstrainNewCodec2.
    • 3edbaaf : Ignore empty data in DataChannel::Send to match FF's behavior.
    • 99f6308 : (Auto)update libjingle 71460499- 71464449
    • a0b929b : Revert "Reland r6707 with the fix for callclient.cc."
    • 196ae6d : (Auto)update libjingle 71456344- 71456420
    • 3dec81a : (Auto)update libjingle 71456173- 71456344
    • a6e8cf8 : Reland r6707 with the fix for callclient.cc.
    • f563e85 : This is to re-open an earlier CL
    • 60e65b1 : (Auto)update libjingle 71452608- 71453580
    • 8636fc8 : Creates the default track if the remote media content is send-only and there is no stream in the SDP.
    • ff50deb : Runtime guard for iOS7 property.
    • 9343cf6 : Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.
    • ba92c52 : Disable GetStats on DrMemory.
    • 026859b : This is related to an earlier CL of enabling Opus 48 kHz. https://webrtc-codereview.appspot.com/16619005/
    • e6f84ae : Initial WebRtcVideoEngine2::GetStats().
    • e9e4253 : Sleep in ThreadTest thread functions.
    • d1ea06b : Restart VideoReceiveStreams in WebRtcVideoEngine2.
    • c31651d : (Auto)update libjingle 71378257- 71410012
    • e364ac9 : AudioBuffer: Optimize const accesses to arrays that autoconvert int16-float
    • c145668 : Reduce runtime of RingBufferTest by a factor of 100.
    • 4f5da03 : Use _numMixedParticipants instead of audioFrameList-size() to determine if there're more than one participants.
    • aa93611 : Connect to the turn server if address cannot be resolved by the browser by using unresolved address. This case is only considered for TCP sockets. P2P layer will assume socket will do the resolve by using a proxy.
    • e5995aa : Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth.
    • e10d28c : fix
    • 8b94e3d : Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.
    • 4065988 : Remove unused ExperimentalNS API in AudioProcessing
    • 2b6bc8d : AudioBuffer: Eliminate the SplitChannelBuffer class
    • 5301b0f : Move additional state into WebRtcVideoSendStream.
    • 2561d52 : Simplify AudioBuffer::mixed_low_pass_data API
    • af93fc0 : AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter
    • 2ade42b : Add unit test for MediaFile WAV file writing
    • 4a472fb : Fixes up rtc so that it compiles on iOS 8 SDK. Adds support for UIInterfaceOrientationUnknown (new with in SDK) and makes it the same as UIInterfaceOrientationPortrait.
    • 52eddec : Revert 6707 "Add support of multiple STUN servers in UDPPort."
    • c56ae63 : r6709 lacks a change in BUILD.gn
    • 74aaf29 : Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filte