For somewhat similar funtionality, GoogCcNetworkController can
be used via GoogCcNetworkControllerFactory.
Bug: webrtc:9586
Change-Id: I298050184513f50c1b9ef5c21b8c9b7a6ca46fd5
Reviewed-on: https://webrtc-review.googlesource.com/c/114543
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26040}
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}
Using GetAudio events from SSRCs without incoming packets doesn't make sense, and should be prevented.
Bug: b/116685514
Change-Id: I48e38bb780549c71cb5f68d370a6819634ad487d
Reviewed-on: https://webrtc-review.googlesource.com/c/114321
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26017}
The removed coded causes problems if the same RTCP packet is forwarded
to the congestion controller multiple times.
Bug: webrtc:10125
Change-Id: I659d8f8f3ce3c643710156fa81176ceeaedd714a
Reviewed-on: https://webrtc-review.googlesource.com/c/114165
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26016}
This reverts commit 0cc42d47389c039c57e47d7ec0c76b97e2da2b0b.
Reason for revert: Sorry, broke WebRTC roll to Chromium again: https://chromium-review.googlesource.com/c/chromium/src/+/1377299. This time the define now set enabled code around the feature flag already landed and there were failures related to that. I suggest to revert that Chromium CL and re-land it after this CL has landed and been rolled into Chromium (if possible to do so).
Original change's description:
> Reland "Default to dlopening the PipeWire."
>
> This is a reland of a099877d8946eb942046ca1295cc142e4fa7ea6f
>
> Original change's description:
> > Reland "Default to dlopening the PipeWire."
> >
> > This is a reland of a13be019017449c57f48203d0fb778f34f7553a7
> >
> > Original change's description:
> > > Default to dlopening the PipeWire.
> > >
> > > Reuse the existing infra from Chromium to do that. Additionally the
> > > target_gen_dir needs to the added to the include directories, otherwise
> > > the Chromium build will fail as it won't find the generated stubs. Also the
> > > pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> > > require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> > > doesn't work with them correctly. With all these changes in place the PipeWire
> > > support is enabled when compiling on Linux.
> > >
> > > Bug: chromium:682122
> > > Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > > Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> > > Cr-Commit-Position: refs/heads/master@{#25720}
> >
> > Bug: chromium:682122
> > Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
> > Reviewed-on: https://webrtc-review.googlesource.com/c/113040
> > Reviewed-by: Weiyong Yao <braveyao@chromium.org>
> > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25981}
>
> Bug: chromium:682122
> Change-Id: Ief26c93069f946f981340664a267fcb412229285
> Reviewed-on: https://webrtc-review.googlesource.com/c/114163
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26004}
TBR=phoglund@webrtc.org,mbonadei@webrtc.org,oprypin@webrtc.org,braveyao@webrtc.org,braveyao@chromium.org,tomas.popela@gmail.com
Change-Id: I9ca52c61210e94182dd6b6a26a136c7f79a2dd0f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:682122
Reviewed-on: https://webrtc-review.googlesource.com/c/114427
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26014}
This is a reland of a099877d8946eb942046ca1295cc142e4fa7ea6f
Original change's description:
> Reland "Default to dlopening the PipeWire."
>
> This is a reland of a13be019017449c57f48203d0fb778f34f7553a7
>
> Original change's description:
> > Default to dlopening the PipeWire.
> >
> > Reuse the existing infra from Chromium to do that. Additionally the
> > target_gen_dir needs to the added to the include directories, otherwise
> > the Chromium build will fail as it won't find the generated stubs. Also the
> > pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> > require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> > doesn't work with them correctly. With all these changes in place the PipeWire
> > support is enabled when compiling on Linux.
> >
> > Bug: chromium:682122
> > Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> > Cr-Commit-Position: refs/heads/master@{#25720}
>
> Bug: chromium:682122
> Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
> Reviewed-on: https://webrtc-review.googlesource.com/c/113040
> Reviewed-by: Weiyong Yao <braveyao@chromium.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25981}
Bug: chromium:682122
Change-Id: Ief26c93069f946f981340664a267fcb412229285
Reviewed-on: https://webrtc-review.googlesource.com/c/114163
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26004}
Without the added preprocessor check, iOS device will be using the desktop logic to determine the number of thread. This put iPhone 8 and iPhone X to use 3 threads and all other iPhones after iPhone 5 to use a single thread.
This CL added a preprocessor for WEBRTC_IOS to have it own thread number calculation logic. In which, the maximum number of thread is fetched from a field_trial and capped by the number of CPU available on the device.
Bug: webrtc:10005
Change-Id: I8c6257fcbf85b07bc986b5f733dbabb3feee37f7
Reviewed-on: https://webrtc-review.googlesource.com/c/110941
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25997}
This CL simplifies a lot of code that can be cleaned up after the merge
of RtpTransportControllerSend and SendSideCongestionController.
In particular, the role of CongestionControlHandler is reduced to only
handle the pacer pushback and stream pausing mechanism.
Bug: webrtc:9586
Change-Id: Idbc1e968efd35e6df6129bc307f6bc1db18d20f2
Reviewed-on: https://webrtc-review.googlesource.com/c/113947
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25994}
- Add chroma siting to color space RTP extension.
- Use 16 bits for max/min luminance.
- Change denominator of chromaticity and luminance.
- Add RTC_DCHECKs to protect against overflows.
Bug: webrtc:8651
Change-Id: If8b95bad6241381224eaba9c5bccce06a65a9195
Reviewed-on: https://webrtc-review.googlesource.com/c/113804
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25990}
This reverts commit a099877d8946eb942046ca1295cc142e4fa7ea6f.
Reason for revert: Breaks WebRTC roll into Chromium. See https://chromium-review.googlesource.com/c/chromium/src/+/1373891:
In file included from ../../third_party/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.cc:11:
In file included from ../../third_party/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.h:16:
../../third_party/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.h:16:10: fatal error: 'pipewire/pipewire.h' file not found
#include <pipewire/pipewire.h>
^~~~~~~~~~~~~~~~~~~~~
Original change's description:
> Reland "Default to dlopening the PipeWire."
>
> This is a reland of a13be019017449c57f48203d0fb778f34f7553a7
>
> Original change's description:
> > Default to dlopening the PipeWire.
> >
> > Reuse the existing infra from Chromium to do that. Additionally the
> > target_gen_dir needs to the added to the include directories, otherwise
> > the Chromium build will fail as it won't find the generated stubs. Also the
> > pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> > require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> > doesn't work with them correctly. With all these changes in place the PipeWire
> > support is enabled when compiling on Linux.
> >
> > Bug: chromium:682122
> > Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> > Cr-Commit-Position: refs/heads/master@{#25720}
>
> Bug: chromium:682122
> Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
> Reviewed-on: https://webrtc-review.googlesource.com/c/113040
> Reviewed-by: Weiyong Yao <braveyao@chromium.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25981}
TBR=phoglund@webrtc.org,mbonadei@webrtc.org,oprypin@webrtc.org,braveyao@webrtc.org,braveyao@chromium.org,tomas.popela@gmail.com
Change-Id: Icdb6a94c8825f13d75ddc12219e99cee8fef51a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:682122
Reviewed-on: https://webrtc-review.googlesource.com/c/114162
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25989}
This is a reland of a13be019017449c57f48203d0fb778f34f7553a7
Original change's description:
> Default to dlopening the PipeWire.
>
> Reuse the existing infra from Chromium to do that. Additionally the
> target_gen_dir needs to the added to the include directories, otherwise
> the Chromium build will fail as it won't find the generated stubs. Also the
> pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> doesn't work with them correctly. With all these changes in place the PipeWire
> support is enabled when compiling on Linux.
>
> Bug: chromium:682122
> Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#25720}
Bug: chromium:682122
Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
Reviewed-on: https://webrtc-review.googlesource.com/c/113040
Reviewed-by: Weiyong Yao <braveyao@chromium.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25981}
Some tests had to be updated due to this change.
Bug: webrtc:9510
Change-Id: I79c4c0166d8ba5e8190a607d5d35b67dc30a3c14
Reviewed-on: https://webrtc-review.googlesource.com/c/113522
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25952}
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.
Original comment from upstream change:
> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.
Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
Changing VAD (voice activity detector) confidence threshold from 40%
to 90%. The proportion of samples classified as speech drops to ca 80%
of what it was when the threshold was 40%. Therefore,
kFullBufferSizeMs has to be increased by 1.0/0.8. We increase it from
1600ms to 2000ms.
TESTED = Did run the new and old configs on AEC dumps. With one minute
of kitchen noise, the new tuning boosted the noise by 3-4 db less.
Bug: chromium:913430
Change-Id: I4a2ebb6d1d309c6c20dd23c3685818b1b5ad4a66
Reviewed-on: https://webrtc-review.googlesource.com/c/113806
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25950}
This change converts all tests but CodecInternalCng and
DecodingErrorDuringInternalCng, which depend on the obsolete Decode
method.
Bug: webrtc:10080
Change-Id: I34b068b3aa7139ed24bd63b417a5adcfc1de7922
Reviewed-on: https://webrtc-review.googlesource.com/c/113506
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25946}
* Stop encoding spatial layers S(n >= N) if application deactivates
spatial layer N by setting RTCRtpEncodingParameters.active = false.
* Move calculation of padding bitrate to SvcRateAllocator class.
* Pad up to minimum required bitrate of base layer if ALR probing is
enabled.
Bug: webrtc:9350
Change-Id: I398284c943d43348def535c83263fc234c9767fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25945}
When screen is zoomed in/out, OSX only updates the parts of Rects currently
displayed on screen, with relative location to current top-left on screen.
This will cause problems when we copy the dirty regions to the captured
frame. So we invalidate the whole screen to copy all the screen contents.
- With CGI method, the zooming will be ignored and the whole screen contents
will be captured as before.
- With IOSurface method, the zoomed screen contents will be captured.
Since we can't know the zooming level and focusing location, so we have
to copy the whole screen region for each frame during rooming. And this
will impact peformance a bit (with IOSurface capturer about 5-10 fps
down on MBP.)
Bug: chromium:911862
Change-Id: Icf123cde4d686ab7ce28fa731bc8dac6925492c8
Reviewed-on: https://webrtc-review.googlesource.com/c/113101
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25936}
Read using capacity() method, write using set_buffer() method. This is
a preparation for making the member private, and renaming it to
capacity_.
Bug: webrtc:9378
Change-Id: I2f96679d052a83fe81be40301bd9863c87074640
Reviewed-on: https://webrtc-review.googlesource.com/c/113520
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25934}
This change is based on a discussion for integrating a new statistic that
measures the delay between the first frame being received and the first frame
being decoded. To enable this in the context of FrameEncryption it makes sense
for packet receive timestamps to be unconditionally recorded.
Bug: webrtc:10105
Change-Id: I6b3b0118121db1fe5d4a4fb16cf5d94341cd2b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/113487
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25931}
- Enable vp9 flexible mode in VideoEngine if 3 spatial layers are set.
- Enable flexible mode in loopback tools and quality tests.
- Reset first active spatial layer on keyframe in encoder.
- Ensure duplicate references are not set by the sender in video header.
- Set references manually for flexible mode in vp9 encoder.
- Delay new activated layers until next base layer frame.
- On receive side put each spatial layer as a separate frame to FrameBuffer
and return several frames combined from FrameBuffer.
Bug: webrtc:10049,webrtc:9794,webrtc:9784
Change-Id: I01e69f134cc145deba666ccc92deb1d37a324ede
Reviewed-on: https://webrtc-review.googlesource.com/c/112289
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25895}
It's currently used only by the VCMJitterBuffer and VCMReceiver
classes. Injection is needed by the VCMReceiverTimingTest test, which
defines a subclass(!) of EventWrapper.
Bug: webrtc:3380
Change-Id: I765be0ceac58e941928319cc426ba49f1cbdc5fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25893}
