Change implementation of `FinalizeFecHeader` to write the FEC header
for multiple ssrcs according to the updated RFC.
Change-Id: I280964b2e53c3579f348fbd42815c966840375ac
Bug: webrtc:15002
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307601
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40270}
Changed FinalizeFecHeader to recieve a list of `ProtectedStream` struct,
in order to prepare for receiving multiple ssrcs to protect in the same
FEC packet header. Implementation of the multistream case will follow in
next CL.
Change-Id: I697ef9172a07797a6f500b9ec3a9916f8f45bc04
Bug: webrtc:15002
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307620
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40269}
This CL adds a new video encode tool that supports to encode video at
specified codec, scalability mode, resolution, frame rate, bitrate,
key frame interval and the number of encoding frames.
The video encoder accepts video frames from `FrameGeneratorInterface`,
and supports `SquareFrameGenerator`, `SlideFrameGenerator` and
`IvfFileFrameGenerator`.
All the encoded bitstreams are wrote into ivf output files.
The purposes of this video encoder tool are:
1. Check the functionalities of video codecs and scalability modes.
2. Optimize video quality at different encode setting.
3. Fine tune the bitrate controller.
4. Compare the quality of different codecs at the same setting.
5. And more.
TESTS: Run the tool at 1280x720, 30fps, 2000kbps, 100 GOP, 300 frames:
vp8 [L1T1 L1T3]
h264 [L1T1 L1T3]
vp9 [L1T1 L1T3 L3T3_KEY]
av1 [L1T1 L1T3 L3T3_KEY]
Bug: webrtc:15210
Change-Id: I3b0e463cf3236cd9a481fbab5688643c203958da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307361
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40267}
into separate sections for each implemented class.
Bug: webrtc:13931
Change-Id: I600f49f3fb195761d13d304f112f36c7c62689df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308120
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40260}
This can be done now as the function SetRTPTimestamp is now overriden
in blink MockTransformableAudioFrame.
Change-Id: I4fa4cb81d0282fea864818f0f2d9a5ed881a5d30
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40257}
VoipCore still use RtpSenderEgress::NonPacedPacketSender, therefore
packets sent using NonPacedPacketSender::EnqueuePackets are proxied
to the worker thead.
When NonPacedPacketSender is used, the Pacer already guarantee that packets are sent on the worker queue.
Lock is removed from RtpSenderEgress and instead calls must be made on
the worker thread.
Bug: webrtc:15209
Change-Id: Iaf03377ad8a037ecedbbe588a4c1e8e4eadacd81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306960
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40252}
Allow absolute send time to go back in time as long as there has not been a large gap in arival time. Use the first packets arival time as time base.
Bug: b/282153758, webrtc:15230
Change-Id: I8663079ab9c202079bf8db303353918d46ba1d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308142
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40251}
Pushing it to the list of extensions to negotiate could result
in enabling it in production.
BUG=None
Change-Id: I98599e9fbac7e2b81b3f2ad0c7759bb052d9d9d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306101
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40250}
To avoid downstream repositories having to deal with a roller breakage
on every major revision to the Android NDK, use an unversioned CIPD
path. Versioned CIPD paths cause downstream roller scripts to assume
each version is an unrelated package and requires manual intervention.
Bug: chromium:1446443
Change-Id: I40e8ecec5a451a1ec754c04b35fa2c26519dd528
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308320
Auto-Submit: Prashanth Swaminathan <prashanthsw@google.com>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Prashanth Swaminathan <prashanthsw@google.com>
Cr-Commit-Position: refs/heads/main@{#40249}
This CL implements {,Logging}DelayVariationCalculator, whose purpose is to calculate simple inter-arrival metrics for a sequence of RTP frames. Uses could include RtcEventLog analysis and ad hoc testing.
Want lgtm: asapersson
Bug: webrtc:15213
Change-Id: I3f9d13a2c4fa66b6f1229c1b6fcd66a6911070de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306741
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40247}
Now that chromium/base has rolled and switched the android_ndk_root to
the new android_toolchain directory, remove the stale Android NDK. Also
update the license generation and build helper scripts to remove
references to the previous NDK.
Bug: chromium:1448383
Test: Verified build of WebRTC.
Change-Id: Ic2b6009f454d67da60231bbcbb5c27bde8407ef3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307541
Commit-Queue: Prashanth Swaminathan <prashanthsw@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40246}
Chromium is being updated to 'android_toolchain', which means the
'android_ndk' DEPS is no longer present. Remove it from the roller until
the transition is complete, then it can be removed from this script
entirely.
Bug: chromium:1448383
Test: Verified fix in libyuv, which uses identical roller script.
Change-Id: I0b07aefaa8edf0a8f87b2accfbcffd4a2307f9f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Prashanth Swaminathan <prashanthsw@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40243}
The luci-analysis system now uses the new threhold schema for bug filing. Remove old fields as they are no longer used anywhere.
Bug: None
Change-Id: Iaa8e7d62ff12fb7e16563c762296efa559e72a92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308060
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Beining Chen <beining@google.com>
Cr-Commit-Position: refs/heads/main@{#40241}
Header metadata such as audio level and capture time doesn't make sense
for redundant payloads (i.e. RED and inband-FEC).
It is assumed that one of the parsed payload timestamps will correspond
to the RTP header timestamp.
This fixes a bug where capture time and CSRCs were not set after
parsing RED packets.
CreateRedPayload test function is adapted from red_payload_splitter_unittest.cc
Bug: webrtc:15185
Change-Id: Iba58772499b6d760f516854999b60511896b053c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305700
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40240}
The Android NDK dependency is moving to a CIPD bucket to reduce the
checkout cost and to eventually move to NDK v25. This introduces the
NDK into an 'android_toolchain' directory. Following the roll of
chromium/base in this repository, a second change will delete the old
'android_ndk' checkout. As a result, the checkout size of this
repository will temporarily increase.
This change also updates the license generation scripts and build
helpers to ensure the android_toolchain is also accounted for.
Bug: chromium:1448383
Test: Verified local builds of WebRTC.
Change-Id: Ibe667be241e5a454d884482061dd10b9850be25f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307540
Commit-Queue: Prashanth Swaminathan <prashanthsw@google.com>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40239}
This change will make it possible to let us modify timestamp in
RTCEncodedAudioFrame.
Change-Id: I97e9571c258fd718d6c211014f1476ca46c78097
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307501
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40238}
Use 8 threads for > 720p
Use 4 tile columns and 2 tile rows for 8 threads
Use 2 tile columns and 2 tile rows for 4 threads
For VGA, 2 tile_col x 2 tile_row configuration has
- ~9% speedup over 4 tile_col x 1 tile_row
- ~5% speedup over 1 tile_col x 4 tile_row
Bug: None
Change-Id: I3c1ea948437aece7c6734ce25351158cbdf0a15b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307880
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40237}
It is not used any more.
Bug: webrtc:13931
Change-Id: I266de41abe239907c6d65f4b182a8dc3aacaba3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308022
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40234}
As per the comment in https://webrtc-review.googlesource.com/c/src/+/303240
on the flexfec_header_reader_writer2.h, renaming this file to flexfec_header_reader_writer.h
and renaming the current implementation to flexfec_03_header_reader_writer.h
as it is based on the 03 draft of the RFC.
Change-Id: I80cb2aba6225ec7cd989a134c3204d1db0ac6f7c
Bug: webrtc:15002
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307600
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40231}
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
