This appears unused. If deleted, other code related to isac bandwidth
estimation becomes unused and may be deleted in followup cls.
Bug: webrtc:10098
Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29252}
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
(a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
sufficient for most production cases.
Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
This CL is a no-op since rtc_use_lto is always false and in general
such change should probably be implemented in
//build/config/compiler/BUILD.gn.
Bug: chromium:408997
Change-Id: Id37d3181e66e699f8cd535aee1af7609352a7259
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149833
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28919}
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}
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}
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}
I will deprecate deps in proto_library for improved build throughput.
We can use link_deps here instead.
Bug: chromium:938011
Change-Id: Iafa83000c3f7f9ffdc0c376a2297b4a9380b7594
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Takuto Ikuta <tikuta@google.com>
Cr-Commit-Position: refs/heads/master@{#26989}
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).
The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.
Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.
[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
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}
Import proto_library.gni when rtc_enable_protobuf is true instead of when
build_with_mozilla is false.
Makes it maybe easier to reason about the intention (e.g. intention is to not
compile any protobuf in, hence flag rtc_enable_protobuf)
The build file could not work if build_with_mozilla = true but
rtc_enable_protobuf = true.
Bug: webrtc:10338
Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597
Reviewed-on: https://webrtc-review.googlesource.com/c/123763
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26868}
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}
In particular, time_utils.h is currently pulled in via rtc_event.h
This CL is in preparation of moving parts of the RTC event log to api/.
Bug: webrtc:10206
Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91
Reviewed-on: https://webrtc-review.googlesource.com/c/117921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26326}
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}
This will print out the major events during a NetEq simulation.
Bug: b/116685514
Change-Id: Iab172e9a9115695b42c67628d5523c727359bb89
Reviewed-on: https://webrtc-review.googlesource.com/c/114320
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26019}
