This reverts commit 90bace0958.
Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.
Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
>
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
>
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
>
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
>
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
>
> TBR=solenberg
>
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}
TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org
Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
This reverts commit 54e41dd08a.
Reason for revert: We are reverting also https://webrtc-review.googlesource.com/c/src/+/16180, so this CL will be included in the re-land of https://webrtc-review.googlesource.com/c/src/+/16180.
Original change's description:
> Remove const from ThreadChecker in NullAudioPoller.
>
> TBR=henrika@webrtc.org,solenberg@webrtc.org
>
> Bug: webrtc:8482
> Change-Id: Ib2738224e776618c692db95cd9473335bc17be15
> Reviewed-on: https://webrtc-review.googlesource.com/17540
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20505}
TBR=terelius@webrtc.org
Change-Id: I27c70ce331043ffdfec676c7e1a51e741d2fe770
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8482
Reviewed-on: https://webrtc-review.googlesource.com/17700
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20511}
This, presumably, will make LUCI tryjobs run for people
opted into the experiment.
Bug: chromium:749455,chromium:776347
Change-Id: I25e3a011810aa47e5f245e1eb0ea2547e29fa3ff
Reviewed-on: https://webrtc-review.googlesource.com/16420
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20506}
uint8_t was being printed as a char; a conversion to int was necessary.
Bug: None
Change-Id: I4c6875c693350b95b8742a6a8e17157743db62cb
Reviewed-on: https://webrtc-review.googlesource.com/17400
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20502}
This reverts commit a7678667fc.
Reason for reland: Fix initializer list constructor.
Original change's description:
> Revert "Add helper functions for migrating to new video codec factories"
>
> This reverts commit 1c9623c70d.
>
> Reason for revert: Breaks brace initialization:
>
> cricket::VideoDecoderParams params = { "deadbeef" };
>
> I suggest adding an initializer list constructor.
>
> Original change's description:
> > Add helper functions for migrating to new video codec factories
> >
> > This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
> > convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
> > webrtc::VideoEncoder and webrtc::VideoDecoder.
> >
> > The purpose is to make it as easy as possible for clients to migrate to the new
> > API and allow us to stop depending on the internal SW codecs as soon as possible.
> >
> > There still exists an ugly decoder adapter class in the video engine. The reason
> > is that we need to continue to pass in the |receive_stream_id| decoder params to
> > some legacy clients.
> >
> > Bug: webrtc:7925
> > Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
> > Reviewed-on: https://webrtc-review.googlesource.com/15181
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20475}
>
> TBR=magjed@webrtc.org,andersc@webrtc.org
>
> Change-Id: I0d1084dc86979fbca748d9ba287d1db3dbe52b44
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7925
> Reviewed-on: https://webrtc-review.googlesource.com/17160
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20486}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,andersc@webrtc.org
Change-Id: Ic825d133b6e1c6e5aad811ba528751dd5ed85e67
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/17360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20501}
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.
This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.
The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.
TBR=solenberg
Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
Intended to enable unit testing of the pacer with a mock PacketQueue.
Bug: webrtc:8422
Change-Id: I142386b2d91ad0d5ba8f3f9d876e67972c490de4
Reviewed-on: https://webrtc-review.googlesource.com/17300
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20498}
Issue was found by the Audio Processing fuzzer.
Bug: chromium:778939, chromium:778921, chromium:778919
Change-Id: If613cf4c533f546d118f10a6358cecd329958177
Reviewed-on: https://webrtc-review.googlesource.com/16161
Commit-Queue: Alex Loiko <aleloi@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20494}
This CL is the step 1 for adding alpha channel support over the wire in webrtc.
- Add the footprint for adapter classes that wraps actual codecs.
- This CL does not add a webrtc::VideoFrame container that can carry alpha to
make the CL shorter for an easier review. Therefore, it exercises a code path
for when we receive no alpha input, just regular I420 frames.
- Unittest sends a video frame for encode/decode through these adapters and
checks the output PSNR.
- See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa
Reviewed-on: https://webrtc-review.googlesource.com/11841
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20490}
This reverts commit 1c9623c70d.
Reason for revert: Breaks brace initialization:
cricket::VideoDecoderParams params = { "deadbeef" };
I suggest adding an initializer list constructor.
Original change's description:
> Add helper functions for migrating to new video codec factories
>
> This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
> convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
> webrtc::VideoEncoder and webrtc::VideoDecoder.
>
> The purpose is to make it as easy as possible for clients to migrate to the new
> API and allow us to stop depending on the internal SW codecs as soon as possible.
>
> There still exists an ugly decoder adapter class in the video engine. The reason
> is that we need to continue to pass in the |receive_stream_id| decoder params to
> some legacy clients.
>
> Bug: webrtc:7925
> Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
> Reviewed-on: https://webrtc-review.googlesource.com/15181
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20475}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: I0d1084dc86979fbca748d9ba287d1db3dbe52b44
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/17160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20486}
The method prototype is being changed to returning a const char*,
we don't rely on it in webrtc and the point of the DCHECK that
previously referenced it, was to avoid usage of std::string for
histogram names.
Bug: webrtc:8472
Change-Id: I69b588d4a8f339911a051fd232d63ea5bb1f9a45
Reviewed-on: https://webrtc-review.googlesource.com/16940
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20484}
Enable cpplint check in the PRESUBMIT for pc/ and fix all existing
warnings.
Bug: webrtc:5583
Change-Id: If39994692ab6f6f3c83c74f23850f02fdfe810e8
Reviewed-on: https://webrtc-review.googlesource.com/16540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20482}
This affects the new injectable codecs.
Bug: webrtc:8459
Change-Id: I484a3ae4c29fd8bae8b13308315758b3689bdd4d
Reviewed-on: https://webrtc-review.googlesource.com/16861
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20478}
This is a reland of 30915a742d
Original change's description:
> Simple Default ObjC video codec factories.
>
> Move the simple video encoder/decoder factory from AppRTCMobile into the
> public API so users who don't have special requirements for video codecs
> can easily get started.
>
> Also clean up the API a little.
>
> This CL replaces the more flexible default factories in
> https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
> want to implement their own codecs will have to supply their own
> encoder/decoder factories as well. The benefits of the approach in
> this CL are a simpler API and less effects on the rest of the code.
>
> Bug: None
> Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
> Reviewed-on: https://webrtc-review.googlesource.com/15141
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20441}
Bug: None
Change-Id: If0910cc540dc835dfec4eeb5bea527d88482d110
Reviewed-on: https://webrtc-review.googlesource.com/16780
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20476}
This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
webrtc::VideoEncoder and webrtc::VideoDecoder.
The purpose is to make it as easy as possible for clients to migrate to the new
API and allow us to stop depending on the internal SW codecs as soon as possible.
There still exists an ugly decoder adapter class in the video engine. The reason
is that we need to continue to pass in the |receive_stream_id| decoder params to
some legacy clients.
Bug: webrtc:7925
Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
Reviewed-on: https://webrtc-review.googlesource.com/15181
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20475}
Adds a unittest to test this.
A Reset() with unsupported frequencies will fail, but currently leaves the resampler in an illegal state.
Subsequent calls to ResetIfNeeded(), which depends on the internal state, will then have unreliable behavior.
The following sequence of calls demonstrate this: It appears as though the resampler is successfully reinitialized to upsample from 44 kHz to 48 kHz, but will in fact crash on Push().
Resampler::Reset() with in=44000, out=32000 // Returns 0
Resampler::ResetIfNeeded() with in=44000, out=48000 // Returns -1
Resampler::ResetIfNeeded() with in=44000, out=48000 // Returns 0
Resampler::Push() with some data
Bug: webrtc:8426
Change-Id: Id1e0528ffcb7a86702d4c2f4c5103a1db419c07d
Reviewed-on: https://webrtc-review.googlesource.com/16424
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20474}
The RTPFragmentationHeader was used when sending audio using RED
for loss protection. This feature has been deprecated and
gradually removed. This cl removes remnants of support from
the RTP send path.
Bug: webrtc:6471
Change-Id: Ia1249047b09c16f79498827f74c2ce07aa38b8f7
Reviewed-on: https://webrtc-review.googlesource.com/16427
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20473}
This CL replaces one 'int32_t' with 'uint32_t'. The value is a
non-negative energy, and the number of leading zeros is
computed. During computation, a shift can cause it to overflow.
Issue was found by the Audio Processing fuzzer.
Bug: chromium:778939, chromium:778921, chromium:778919
Change-Id: I3d7e0b547e6b0edcd9995903517ea851142a08c1
Reviewed-on: https://webrtc-review.googlesource.com/16433
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20470}
