This reverts commit 24192c267a.
Reason for revert: Analyzed the performance regression in more detail.
Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f6 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac.
There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall.
Original change's description:
> Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
>
> This reverts commit 3e8ef940fe.
>
> Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260.
>
> Original change's description:
> > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
> >
> > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
> >
> > Bug: webrtc:10668
> > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > Commit-Queue: Chen Xing <chxg@google.com>
> > Cr-Commit-Position: refs/heads/master@{#28434}
>
> TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com
>
> Bug: webrtc:10668, chromium:982260
> Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28561}
TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10668, chromium:982260
Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28664}
This reverts commit 3e8ef940fe.
Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260.
Original change's description:
> Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
>
> This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
>
> Bug: webrtc:10668
> Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Chen Xing <chxg@google.com>
> Cr-Commit-Position: refs/heads/master@{#28434}
TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com
Bug: webrtc:10668, chromium:982260
Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28561}
This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28434}
This is a reland of 0ded32d5a3
Original change's description:
> Reland "Change buffer level filter to store current level in number of samples."
>
> This is a reland of 87977dd06e
>
> Original change's description:
> > Change buffer level filter to store current level in number of samples.
> >
> > The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
> >
> > Bug: webrtc:10736
> > Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28368}
>
> Bug: webrtc:10736
> Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28393}
Bug: webrtc:10736
Change-Id: I251b8321e5a5fd870e018bc7c8083ec0a41de81b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144023
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28398}
This reverts commit 0ded32d5a3.
Reason for revert: breaks downstream projects.
Original change's description:
> Reland "Change buffer level filter to store current level in number of samples."
>
> This is a reland of 87977dd06e
>
> Original change's description:
> > Change buffer level filter to store current level in number of samples.
> >
> > The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
> >
> > Bug: webrtc:10736
> > Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28368}
>
> Bug: webrtc:10736
> Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28393}
TBR=henrik.lundin@webrtc.org,minyue@webrtc.org,jakobi@webrtc.org
Change-Id: I570c83ec3a88a24d7a1f883a351748dd71bea015
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144022
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28397}
This is a reland of 87977dd06e
Original change's description:
> Change buffer level filter to store current level in number of samples.
>
> The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
>
> Bug: webrtc:10736
> Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28368}
Bug: webrtc:10736
Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28393}
This reverts commit 87977dd06e.
Reason for revert: Breaks downstream project
Original change's description:
> Change buffer level filter to store current level in number of samples.
>
> The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
>
> Bug: webrtc:10736
> Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28368}
TBR=henrik.lundin@webrtc.org,minyue@webrtc.org,jakobi@webrtc.org
Change-Id: I3900c9f6071fce51d13fb3b7c886157304d7a5c3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143786
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28369}
The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
Bug: webrtc:10736
Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28368}
Several new audio stats were recently standardized and implemented in
WebRTC in https://webrtc-review.googlesource.com/c/src/+/133887. This CL
adds these to the GetStats API.
Bug: webrtc:10442, webrtc:10443, webrtc:10444
Change-Id: I0e898ac14777e82b1a9099b5e0a5584eb9cb5934
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134213
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27839}
The metrics are now added as RTCNonStandardStatsMember objects in
RTCMediaStreamTrackStats. Unit tests are updated.
This is a reland of https://webrtc-review.googlesource.com/c/src/+/134303,
with fixes.
TBR=kwiberg@webrtc.org
Bug: webrtc:10549
Change-Id: I29dcc6fbfc69156715664e71acfa054c1b2d9038
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134500
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27806}
It appears unused everywhere. It will be deleted in a followup cl.
Bug: webrtc:6471
Change-Id: Ief992db6e52aee3cf1bc77ffd659ffbc072672ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134212
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27787}
Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely.
Adds an ENCODER and sets it up to parse SDP config for multistream
opus.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27775}
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.
This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
Switch to explicit channel mappings (RFC 7845) when creating
multi-stream Opus en/de-coders. The responsibility of setting up the
channel mappings will shift from WebRTC to the WebRTC user.
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
current vision. See also the first child CL
https://webrtc-review.googlesource.com/c/src/+/129768
that sets up the Decoder to use this code.
Bug: webrtc:8649
Change-Id: I55959a293d54bb4c982eff68ec107c5ef8666c5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129767
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27452}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
The difference to the original is new bitexactness strings. The
reason for reland is breaking downstream projects.
Original CL description:
Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
TBR=ossu@webrtc.org
Bug: webrtc:8649
Change-Id: I6261b18c69fd666d43ab34ed8f1bc9d5cc82b21f
Reviewed-on: https://webrtc-review.googlesource.com/c/123882
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26809}
This reverts commit 5341aaccdb.
Reason for revert: Order of initialization of global static strings.
Original change's description:
> Reland of https://webrtc-review.googlesource.com/c/src/+/114883
>
> The difference to the original is new bitexactness strings AND
> global static file string constants. The reason for reland is breaking
> downstream projects.
>
> Original CL description:
>
> Tests for multi-stream Opus.
>
> This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
> tests are in audio_coding_unittest.cc. Some refactoring of
> AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
> possible. A few checks for "channels \in {1, 2}" are replaced with
> "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
> other changes are made to be able to write and read multi-channel WAV
> files.
>
> The SDP changes are NOT included; as of this CL there is no way to set
> up a multi-stream opus en/de-coder from SDP strings.
>
> Bug: webrtc:8649
> Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
> Reviewed-on: https://webrtc-review.googlesource.com/c/123387
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26774}
TBR=aleloi@webrtc.org,ossu@webrtc.org
Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123580
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26777}
The difference to the original is new bitexactness strings AND
global static file string constants. The reason for reland is breaking
downstream projects.
Original CL description:
Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
Bug: webrtc:8649
Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/123387
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26774}
This reverts commit 9c31ac2323.
Reason for revert: Breaks downstream project.
Original change's description:
> Tests for multi-stream Opus.
>
> This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
> tests are in audio_coding_unittest.cc. Some refactoring of
> AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
> possible. A few checks for "channels \in {1, 2}" are replaced with
> "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
> other changes are made to be able to write and read multi-channel WAV
> files.
>
> The SDP changes are NOT included; as of this CL there is no way to set
> up a multi-stream opus en/de-coder from SDP strings.
>
> Bug: webrtc:8649
> Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c
> Reviewed-on: https://webrtc-review.googlesource.com/c/114883
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26742}
TBR=aleloi@webrtc.org,ossu@webrtc.org
Change-Id: I0ac48b7320d31d3e7af33bf8714c3db6c807b82e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/123385
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26747}
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.
The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.
Bug: webrtc:8649
Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c
Reviewed-on: https://webrtc-review.googlesource.com/c/114883
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26742}
This prepares for making the Clock interface fully mutable.
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
NetEq currently only passes `jitterBufferDelay` to `getStats()`. We need its paired `jitterBufferEmittedCount` denominator stat for the calculations to be accurate.
Bug: webrtc:10192
Change-Id: I655aea629026ce9101409c2e0f18c2fa57a1c3ab
Reviewed-on: https://webrtc-review.googlesource.com/c/117320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#26276}
A trimmed down version is moved to legacy_encoded_audio_frame_unittest.cc
where it's used for test parameterization.
Bug: webrtc:10185
Change-Id: I9abda22f9806b831b6ca4b27d6bcc888285f50f2
Reviewed-on: https://webrtc-review.googlesource.com/c/116961
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26218}
The following APIs on AudioCodingModule are deprecated with this CL:
static int NumberOfCodecs();
static int Codec(int, CodecInst*);
static int Codec(const char*, CodecInst*, int, size_t);
static int Codec(const char*, int, size_t);
absl::optional<CodecInst> SendCodec() const;
bool RegisterReceiveCodec(int, const SdpAudioFormat&);
int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
int UnregisterReceiveCodec(uint8_t);
int32_t ReceiveCodec(CodecInst*);
absl::optional<SdpAudioFormat> ReceiveFormat();
As well as this method on RtpRtcp module:
int32_t RegisterSendPayload(const CodecInst&);
Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}
The stat will be exposed through origin trial described in:
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI
Change-Id: Ib191a11c6bd9e617abbe9dd82239b0c5b4e6b4e0
Bug: webrtc:10043
Reviewed-on: https://webrtc-review.googlesource.com/c/111922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25802}
Expected checksums depend on whether libopus is built with SSE or not.
Since we have no robust way to know that and we cannot enforce all
clients to use SSE, we accept both results.
Bug: webrtc:9530
Bug: webrtc:9995
Change-Id: I9f0464ffec15df91eafe15d89c61e2140f341cb1
Reviewed-on: https://webrtc-review.googlesource.com/c/110789
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25633}
We want to get rid of the whole thing, really, but these two were
easy.
Bug: webrtc:8396
Change-Id: I9292bf077caca171c9f5fe63695b6333cf22547d
Reviewed-on: https://webrtc-review.googlesource.com/c/104763
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25618}
After this CL, all audio encoders have to be injected by the caller.
This means that there is no special "built-in" set of codecs, and
users won't have to pay the binary size and security costs of codecs
they aren't using.
Bug: webrtc:8396
Change-Id: Idb0959ce395940c8bb3bbb49256cdcd84fc87bb6
Reviewed-on: https://webrtc-review.googlesource.com/c/103821
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25600}
And put codecs/cng/webrtc_cng.h in a non-public build target while
we're at it.
Bug: webrtc:8396
Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c
Reviewed-on: https://webrtc-review.googlesource.com/c/108782
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25486}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: I1d7c62fbc2585233cf1656fdcc4bb5380c2f41a5
Reviewed-on: https://webrtc-review.googlesource.com/c/100980
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24947}
It will soon lose the ability to do so.
Bug: webrtc:8396
Change-Id: Ifca101fce0c349dba8c402ab2b6ad614061a88f6
Reviewed-on: https://webrtc-review.googlesource.com/101281
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24836}
Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
The new way of creating encoders used a 32 kbit/s bitrate
unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz
and 56 kbit/s for 32 kHz, which is what the old way of creating
encoders has used since forever.
I also had to change some test expectations on Opus, because the new
way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I
believe to be correct), while the old way defaults to 64 kbit/s in
both cases.
Bug: webrtc:8396
Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2
Reviewed-on: https://webrtc-review.googlesource.com/94540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24375}
This is a reland of 80c4cca491
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
Bug: webrtc:9421
Change-Id: Ice351b635788167f2971b26470f73a5e5fa1a240
Reviewed-on: https://webrtc-review.googlesource.com/86543
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23799}
This reverts commit 80c4cca491.
Reason for revert: Breaks downstream tests.
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
>
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
>
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
> no longer be reached.
>
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}
TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org
Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
The playout modes other than Normal have not been reachable for a long
time, other than through tests. It is time to deprecate them.
The only meaningful use was that Fax mode was sometimes set from
tests, in order to avoid time-stretching operations (accelerate and
pre-emptive expand) from messing with the test results. With this CL,
a new config is added instead, which lets the user specify exactly
this: don't do time-stretching.
As a result of Fax and Off modes being removed, the following code
clean-up was done:
- Fold DecisionLogicNormal into DecisionLogic.
- Remove AudioRepetition and AlternativePlc operations, since they can
no longer be reached.
Bug: webrtc:9421
Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
Reviewed-on: https://webrtc-review.googlesource.com/84123
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23704}
By removing it we can in turn (next CL) get rid of RTPTypeHeader, which is a
union that cause some problems.
Bug: none
Change-Id: I9246ecbfe2c8b7eda27497cccbc5f438958b64bf
Reviewed-on: https://webrtc-review.googlesource.com/83985
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23666}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
NetEq tapers down the audio produced through loss concealment when the
expansion has been going on for some time. When the audio packets starts
coming in again, there is a ramp-up that happens. This ramp-up could
before this change extend over more than one 10 ms block, which made
keeping track of the scaling factor necessary. With this change, we make
this ramp-up quicker in the rare cases when it lasted more than 10 ms,
so that it always ramps up to 100% within one block. This way, we can
remove the mute_factor_array.
This change breaks bit-exactness, but careful listening could not reveal
an audible difference.
This change is a part of a larger refactoring of NetEq's PLC code.
Bug: webrtc:9180
Change-Id: I4c513ce3ed8d66f9beec2abfb1f0c7ffaac7a21e
Reviewed-on: https://webrtc-review.googlesource.com/77180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23342}
Instead of checking for an exact bitrate check that the bitrate is between
the min and max values.
Also relax a threshold in a bandwith adaptation test.
Bug: webrtc:9280
Change-Id: I465d785a53759f73242198ee1ccd7da1a26c48b7
Reviewed-on: https://webrtc-review.googlesource.com/78041
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23330}
This is a reland of 79ded653fe
Original change's description:
> Update expected bitrate in Opus tests
>
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
>
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}
TBR=henrik.lundin@webrtc.org
Bug: webrtc:9280
Change-Id: I6bfcd1c5e1d5298543024a0faa6a695026434df3
Reviewed-on: https://webrtc-review.googlesource.com/77980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23318}
This reverts commit 79ded653fe.
Reason for revert: Different repos have different Opus
Original change's description:
> Update expected bitrate in Opus tests
>
> Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
> CL re-enables recently disabled unittests and updates the expected bitrates.
>
> Bug: webrtc:9280
> Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
> Reviewed-on: https://webrtc-review.googlesource.com/77766
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23306}
TBR=henrik.lundin@webrtc.org,gustaf@webrtc.org
Change-Id: I3c18db2d6052c4049d836c3e595b00189aebcbc8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9280
Reviewed-on: https://webrtc-review.googlesource.com/77800
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23307}
Upstream changes to Opus DTX behavior changes the bitrates of Opus. This
CL re-enables recently disabled unittests and updates the expected bitrates.
Bug: webrtc:9280
Change-Id: I668a0b6a8b82cbbb70d795db4546cb5469266bf2
Reviewed-on: https://webrtc-review.googlesource.com/77766
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23306}
In order to roll the new version of Opus in WebRTC, this CL disables
some tests that will fail because of [1].
They will be re-enabled and fixed as soon as the new Opus revision is
rolled.
[1] - https://chromium-review.googlesource.com/1061499TBR=henrik.lundin@webrtc.org
Bug: webrtc:9280
Change-Id: I84870ced66d554f75c2d093dac8103ad7860cae5
Reviewed-on: https://webrtc-review.googlesource.com/77640
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23293}
Only half of the array was initialized. Now all of it is.
Bug: chromium:839960
Change-Id: If8bbe12c4c4c0dc0d529c93b22e49a94ecb09919
Reviewed-on: https://webrtc-review.googlesource.com/74820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23167}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.
BUG=webrtc:5801, webrtc:8396
Change-Id: I80749ec3b86cba73994307046d05964f59167d44
Reviewed-on: https://webrtc-review.googlesource.com/18440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22774}
This prepares us for removing them altogether.
Bug: webrtc:8982
Change-Id: I66002cc8d4bf0e07925766d568d2498422f0f38e
Reviewed-on: https://webrtc-review.googlesource.com/64142
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22707}
MockAudioEncoder was calling a mocked Die function on itself in its
destructor. This outputs "Uninteresting mock function call" warning if
the Die call was not expected. This is true even if a NiceMock is used
to suppress the warnings.
The purpose of testing that the destructor is called might be to protect
against memory leaks when audio encoder ownership is transferred using a
raw pointer. However, this case is already covered by msan checks.
Bug: None
Change-Id: I0603c417b4b239027859228e05ebcf83ff5aaf18
Reviewed-on: https://webrtc-review.googlesource.com/56183
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22146}
This allows removing the specific classes in a later CL.
Bug: webrtc:8677
Change-Id: I3b9c1f3191c38e6d31a3de990e2d882505e79adc
Reviewed-on: https://webrtc-review.googlesource.com/35040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21412}
The UMA metric will log the same information that goes into the
googPreferredJitterBufferMs stat.
Bug: webrtc:8488
Change-Id: I4e4e1e362dd42377105d52d2c4cd49c1ecb1a90d
Reviewed-on: https://webrtc-review.googlesource.com/26740
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20923}
Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=kwiberg@webrtc.org
Bug: None
Change-Id: I055411a3e521964c81100869a197dd92f5608f1b
Reviewed-on: https://webrtc-review.googlesource.com/23619
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20728}
We've done this previously with the other audio encoders, but Opus had
to wait until all external users had been updated.
BUG=webrtc:7847
Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3
Reviewed-on: https://webrtc-review.googlesource.com/8000
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20424}
This is in preparation for deleting the include in rtc_base/refcount.h,
but that change has to wait for some downstream applications to
not rely in the indirect include.
Partial reland of "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
This is a reland of b7239a9dc8
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I63a42712f6c1ec83823c629d1a954fd1a04d4a6c
Reviewed-on: https://webrtc-review.googlesource.com/7281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20185}
This is a reland of b7239a9dc8
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I9738f6680ab52d0f43639a1a39175fdba5957681
Reviewed-on: https://webrtc-review.googlesource.com/5840
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20180}
This reverts commit b7239a9dc8.
Reason for revert: Broke chromium mac build, compilation failures on content/renderer/media/webrtc/webrtc_video_frame_adapter.h.
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
TBR=kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I7334597cc8979ba9cfaff526967084349ef27f3c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8270
Reviewed-on: https://webrtc-review.googlesource.com/5800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20107}
The refcount.h file doesn't depend on anything from
refcountedobject.h. The motivation of this change to make it possible
to add additional declarations to refcount.h, and include it from
refcountedobject.h.
Bug: webrtc:8270
Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
Reviewed-on: https://webrtc-review.googlesource.com/5760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20106}
Description of this stat can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay
Bug: webrtc:8281
Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023
Reviewed-on: https://webrtc-review.googlesource.com/3381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20069}
This method is no longer in use.
Bug: webrtc:3520
Change-Id: Ie1419fa95e6ef482e9adc1ed7af57af2c3510a65
Reviewed-on: https://webrtc-review.googlesource.com/4667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20047}
The number of concealment events. This counter increases every time a concealed sample is
synthesized after a non-concealed sample. That is, multiple consecutive concealed samples
will increase the concealedSamples count multiple times but is a single concealment event.
Bug: webrtc:8246
Change-Id: I7ef404edab765218b1f11e3128072c5391e83049
Reviewed-on: https://webrtc-review.googlesource.com/1221
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19881}
Set limits of NetEq minimum and maximum delay to 0-10000 ms closed interval.
Fixed error message in Audio Coding Module.
Bug: webrtc:6861
Change-Id: Id1b9928f808bdb6e1088c6895f2ec4a53b00efb2
Reviewed-on: https://webrtc-review.googlesource.com/1343
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19860}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
