This is a reland of 0a88ea050c.
The new stat will not be reported unless it is GT 0.
Reporting of decoding_codec_plc events
Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`.
Bug: webrtc:10739
Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28790}
This reverts commit fab3460a82.
Reason for revert: fix downstream instead
Original change's description:
> Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
>
> This reverts commit 9973933d2e.
>
> Reason for revert: breaking downstream projects and not reviewed by direct owners
>
> Original change's description:
> > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
> >
> > 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}
>
> TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
>
> Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10668, chromium:982260
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28671}
TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10668, chromium:982260
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28672}
This reverts commit 9973933d2e.
Reason for revert: breaking downstream projects and not reviewed by direct owners
Original change's description:
> Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
>
> 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}
TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10668, chromium:982260
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28671}
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 is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.
Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
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}
The functionality is hidden behind field trial for experimentation.
Bug: webrtc:10736
Change-Id: I1daf60966717c3ea43bf6ee16d190290ab740ce7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144059
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28474}
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}
The last run-time logic for selecting function pointers was removed in
May 2016, here: https://codereview.webrtc.org/1955413003
It would be even better if we could eliminate the function pointers
entirely and just have different implementations that we select at
compile time; I've left a TODO asking for this.
Bug: webrtc:9553
Change-Id: Ica71d71e19759da00967168f6479b7eb8b46c590
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144053
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28414}
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}
This reverts commit 79890ef91f.
Reason for revert: the sync buffer was actually not counted when the buffer level filter was updated since the value was rounded down to the closest whole packet.
Original change's description:
> Remove sync buffer length from FilteredCurrentDelayMs.
>
> The sync buffer length is already added when the buffer level filter is updated.
>
> Bug: webrtc:10736
> Change-Id: Icbd411d4fd7b16f31b800142d1b6a8de79365d91
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140950
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28261}
TBR=minyue@webrtc.org,jakobi@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10736
Change-Id: Ibf4ce566484ff01421b186e03fe97fe633ba066d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143167
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28335}
The sync buffer length is already added when the buffer level filter is updated.
Bug: webrtc:10736
Change-Id: Icbd411d4fd7b16f31b800142d1b6a8de79365d91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140950
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28261}
It's easy to make small errors when building field trial strings, and
those errors can cause all sorts of weird problems. This CL checks if
the FT string has an odd number of delimiters, duplicate
names or any trailing chars.
If so we'll log a error message. On debug builds we'll also crash.
Bug: webrtc:10729
Change-Id: Iebf7155d9b117a02d1e9cfe7f64408e11df2aec5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140866
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28234}
While reading inpùt files until their end, the assert should be
ASSERT_TRUE.
Change-Id: Ib60b68173b58b77d9789c544c7cb647a752a24d1
Bug: webrtc:10690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140280
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28202}
Insert first packet before calling to decode.
Bug: webrtc:10690
Change-Id: I721b7af0506f0dbaf4fa2ed6a9ba6a87250d08f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139103
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28149}
This is an usual error while using neteq_quality_test. This tool
does not support wav files as input. Adding a validation.
Bug: webrtc:10690
Change-Id: I18ed308d2f688106728df5df25e0a58c7170f411
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139104
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28141}
Instead of setting a runtime, allow neteq_quality_test to
consume a complete file using --runtime_ms -1
Bug: webrtc:10690
Change-Id: I90d35cf31996d9336fef817b9332a2cd1d04e77e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139101
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28134}
In addition to the 48 kHz that we've always used.
Bug: webrtc:10631
Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28104}
Change the way the forget factor converge to the steady state so that we don't overemphasize the first packets received.
The logic is controlled by the delay histogram field trial which has an added parameter to control if emphasis should be even (c=1, default) or put on later packets (c>1) until we reach our steady state forget factor.
Bug: webrtc:10411
Change-Id: Ia5d46c22d1a4a66994652f71c8cde664362bfacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137050
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28039}
This change allows NetEq to reach preferred jitter buffer size much faster
for high target delays because it uses absolute units instead of relative ones
during computation of lower_limit.
More details can be found here:
https://docs.google.com/document/d/12qPMJYFhGXrA_o_nvz9VshpzAJX6aULxFig1fTzBzDI/edit
Change-Id: I21ce0e35e25166d935fdf0325c083bcf990899f5
Bug: webrtc:10619
Change-Id: I21ce0e35e25166d935fdf0325c083bcf990899f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135745
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#27970}
Going back to a ratio in [0.0, 1.0] instead of a % number. Also changed
the format of the tag to match the others.
Bug: webrtc:10549
Change-Id: I03216718156843e345f8d0a76258a15f1a355fbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135104
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27840}
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}
This CL adds a new metric to NetEq, which logs whenever a loss
concealment event has lasted longer than 150 ms (an "interruption").
The number of such events, as well as the sum length of them, is kept
in a SampleCounter, which can be queried at any time.
Any initial PLC at the beginning of a call, before the first packet is
decoded, is ignored.
Unit tests and piping to neteq_rtpplay are included.
Bug: webrtc:10549
Change-Id: I8a224a34254c47c74317617f420f6de997232d88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132796
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27781}
Several new audio stats have been added to the standard, and this CL
implements those inside of NetEq. Exposing these metrics on the API will
be done in a follow-up CL.
Bug: webrtc:10442, webrtc:10443, webrtc:10444
Change-Id: Ia7aa5a6d76685fc0fdb446172a0a3fd0310f6cb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133887
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27755}
Instead of crashing when encountering an event log that cannot be parsed
it is better to print an error message, skip the file and continue.
Bug: webrtc:10337
Change-Id: I5dbca18e456c14e5a92af068f82e88cb17e8de9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133185
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27727}
We knew that we should not update buffer level during DTX period. We already fulfill this upon no packet receipt. But we missed doing it for DTX-signaling packets. This CL is to fix that.
Bug: b/129521878
Change-Id: I72ca18e3b21e956123fe6e3119ef0d7c981c9eec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133183
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27643}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
Currently the code in NetEqTestFactory will crash when something
unexpected happens. It would be better to return a nullptr instead and
let the caller decide how to proceed.
Bug: webrtc:10337
Change-Id: I3cfdffa7e6f2016eeaa5d6e80c5dd6c954ef8485
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127894
Reviewed-by: Pablo Barrera González <barrerap@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27226}
Previously only reading from the filesystem was supported, this CL
allows parsing an event log from a string.
Bug: webrtc:10337
Change-Id: Iadde3319eb8fb4175625f510201fac9c01c80ed9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127296
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27202}
When we offset the measured inter-arrival time due to packet loss, it will sometimes be less than zero. This is the correct value to use when calculating the relative packet arrival delay.
Bug: webrtc:10333
Change-Id: I14a68563a379fa0b9444684304362503a6f1bfca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127547
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27164}
This reverts commit c4b391a257.
Reason for revert: issue fixed
Original change's description:
> Revert "NetEQ RTP Play: Optionally write output audio file"
>
> This reverts commit 6330818ec8.
>
> Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.
>
> Original change's description:
> > NetEQ RTP Play: Optionally write output audio file
> >
> > This CL makes the output audio file optional to more
> > quickly run neteq_rtpplay when no audio output is needed.
> > The CL also includes necessary adaptations because of pre-existing
> > dependencies (e.g., the output audio file name is used to create
> > the plotting script file names).
> >
> > The command line arguments are retro-compatible - i.e., same behavior
> > when specifying the output audio file and the new flag
> > --output_files_base_name is not used.
> >
> > This CL also includes a test script with which the retro-compatibility
> > has been verified.
> >
> > Bug: webrtc:10337
> > Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27067}
>
> TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
>
> Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10337
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27078}
TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
Change-Id: Ia7061f7c2d69db61638ad612e82cd429eb49d539
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27106}
This reverts commit 6330818ec8.
Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.
Original change's description:
> NetEQ RTP Play: Optionally write output audio file
>
> This CL makes the output audio file optional to more
> quickly run neteq_rtpplay when no audio output is needed.
> The CL also includes necessary adaptations because of pre-existing
> dependencies (e.g., the output audio file name is used to create
> the plotting script file names).
>
> The command line arguments are retro-compatible - i.e., same behavior
> when specifying the output audio file and the new flag
> --output_files_base_name is not used.
>
> This CL also includes a test script with which the retro-compatibility
> has been verified.
>
> Bug: webrtc:10337
> Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27067}
TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27078}
This CL makes the output audio file optional to more
quickly run neteq_rtpplay when no audio output is needed.
The CL also includes necessary adaptations because of pre-existing
dependencies (e.g., the output audio file name is used to create
the plotting script file names).
The command line arguments are retro-compatible - i.e., same behavior
when specifying the output audio file and the new flag
--output_files_base_name is not used.
This CL also includes a test script with which the retro-compatibility
has been verified.
Bug: webrtc:10337
Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27067}
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}
This is a reland of d9f798a6b3
Original change's description:
> Remove field trial include from decision logic.
>
> Bug: webrtc:9289
> Change-Id: I2e465bf9eddda8bde50daeb14cfd51405e536ff4
> Reviewed-on: https://webrtc-review.googlesource.com/c/125097
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26925}
Bug: webrtc:9289
Change-Id: I40fbd999fc8495beaeb46799c333f91d72b5be37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125720
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26978}
Previously, if more than one packet is extracted in a GetAudio call then
an incorrect number of samples will be reported.
Bug: webrtc:10363
Change-Id: Ia1bcc87a0e0082060e4f746d37a4008735eec6b3
Reviewed-on: https://webrtc-review.googlesource.com/c/124829
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26903}
This replaces the use of command-line flags with the use of a config
struct. This makes it easier for non command-line applications to use
the NetEqTestFactory to run simulations.
Bug: webrtc:10337
Change-Id: I24533bf206e70e12db9af8d9675769c1ff7c7d48
Reviewed-on: https://webrtc-review.googlesource.com/c/123600
Reviewed-by: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26887}
- This mode estimates relative packet arrival delay for each incoming packet and adds that value to the histogram.
- The histogram buckets are 20 milliseconds each instead of whole packets.
- The functionality is enabled with a field trial for experimentation.
Bug: webrtc:10333
Change-Id: I8f7499c56802fc1aa1ced2f5310fdd2ef1403515
Reviewed-on: https://webrtc-review.googlesource.com/c/123923
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26871}
For e.g. when audio receiver is recreated during SetRtpExtensionsAndRecreateStream in webrtc_voice_engine.h,
the audio minimum delay can't go down.
Imagine we set base minimum playout delay when audio receiver stream is created, then its value will be cached, to be applied during recreation. Then SetRtpExtensionsAndRecreateStream is fired, and audio receiver stream is recreated with the cached value, but currently it in the constructor it is used to initialize both base minimum playout delay and minimum playout delay. Which leads to the bug that effective minimum playout delay can't go down anymore as if you set base minimum playout delay to the low value then effective delay use the biggest value which minimum playout delay.
This didn't come up during previous trials because of
https://webrtc-review.googlesource.com/c/src/+/122280
It was reseting minimum playout delay to 0 asynchronously, that is why you couldn't see this bug.
Bug: webrtc:10287
Change-Id: I924446bfcb33ac94f7e5bf987a1868acaf1b0346
Reviewed-on: https://webrtc-review.googlesource.com/c/124000
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26832}
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}
On NetEq level latency corresponds to delay and two terms can be used interchangeably here.
In order to implement latency constraint we need to provide a range of possible values which should be constant. See getCapabilities() here:
https://www.w3.org/TR/mediacapture-streams/#dfn-applyconstraints-algorithm
Lowest possible value accepted value is constant and equals 0. But because |packet_len_ms_| and |maximum_delay_ms_| may be updated during live of DelayManager upper bound is not constant. Moreover, due to change in |packet_len_ms_| the |minimum_delay_ms_| which was valid when its was set may be considered invalid later on.
To circumvent that and provide constant range for capabilities we separate base minimum delay and minimum delay. ApplyConstraints algorithm will set base minimum delay. Base minimum delay will act as recommendation for lower bound of minimum delay and will be used to limit target delay. If user sets base minimum delay through ApplyConstraints which is bigger than currently
possible maximum (e.g. bigger than NetEq maximum buffer size in milliseconds) then base minimum delay will be clamped to currently possible maximum to match user's intentions as best as possible.
Note, that we keep original behavior when minimum_delay_ms_ (effective_minimum_delay_ms after this CL) in LimitTargetLevel method may be above upper bound due to changing packet audio length.
Bug: webrtc:10287
Change-Id: I06b8f5cd3fd1bc36800af0447f91f7c4dc21a766
Reviewed-on: https://webrtc-review.googlesource.com/c/121700
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26666}
Currently, if the last packet was reordered (e.g. due to retransmission) then the next packet's inter-arrival time will be estimated incorrectly due to the jump in sequence numbers. This change prevents that by not resetting the stopwatch on reordered packets.
This will also better estimate inter-arrival times when we have multiple reordered packets in a burst. Currently we would only measure the iat of the first reordered packet correctly and not the ones coming after it.
There is a slight risk introducing this: If we would receive an out of order packet far into the future (in sequence numbers) and then continue getting packets in the normal order, then we would not update the current sequence number for these and incorrectly estimate their inter-arrival times since they would all be considered reordered.
Change-Id: Ic938a37cbddf1cb9c30b610218f56794568d3d01
Bug: webrtc:10178
Reviewed-on: https://webrtc-review.googlesource.com/c/119949
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26572}
This is first step to allow to set latency
from client code in Chromium.
Existing minimum latency hasn't been used because it can clash
with video syncronization code.
Bug: webrtc:10287
Change-Id: Ia38906506069a1abfa01698dc62df283fc15cfbc
Reviewed-on: https://webrtc-review.googlesource.com/c/121423
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26536}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
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}
When enabled, the delay manager is updated with reordered packets. It also makes the peak detector ignore the reordered packets.
Change-Id: I2bdc99764cc76b15e613ed3dc75f83aaf66eee4e
Bug: webrtc:10178
Reviewed-on: https://webrtc-review.googlesource.com/c/116481
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26187}
This fixes a bug where we sometimes extract an Opus CNG packet and the packet after, even though there was big timestamp gap between the packets, which causes expansion during the next GetAudio calls.
Change-Id: I2409ac08df58afc496f74b91981657b7206e8bb1
Bug: webrtc:10167
Reviewed-on: https://webrtc-review.googlesource.com/c/115419
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26179}
