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}
Later CLs will switch to these functions, and eventually the
CreateMediaChannel will be deprecated and removed.
Bug: webrtc:13931
Change-Id: I4c5ab89659a47a501728cac217bb1a877fa50047
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307800
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40221}
Empty codec objects do not make sense. Instead of creating an empty
object to be used as a placeholder in the API, at least create a
video codec with the right name.
Bug: webrtc:15214
Change-Id: I705d9d1361f353fe5dc538a6fe972c8a346f1247
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307221
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40218}
Most of the usage of the H264Encoder::Create(codec) method passes a
simple codec with just the H264 codec name. This simplified the call
sites in many places and removes references to the codec types.
Bug: webrtc:15214
Change-Id: I4039c0be4ce6e3147c14c7853df4635f344b7d70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307222
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40214}
Part 1 of the migration towards merging the types.
Any method that could belong to the Codec type was moved, the others
are deprecated.
Alternatives to the AudioCodec and VideoCodec constructors are introduced
to allow creating objects of an indefinite type without having to
reference the old classes.
Bug: webrtc:15214
Change-Id: I20e1aa32962821cad98e9a92c2ec86f8f75e5dd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40213}
This ensures that the MediaChannel interface is only implemented
through a send/receive shim, splitting channels also when kBoth is
used.
Bug: webrtc:13931
Change-Id: Ie97809597eaae7b4f504939339795432c34e56cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307461
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40210}
