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 to follow chromium.
4541db8796
Also this is necessary to switch swarming client for webrtc builder not to cause unintentional timeout.
Bug: chromium:894045, chromium:914164
Change-Id: I98d8b6b6d31c5bfbf176e373c1e189a5eadc2838
Reviewed-on: https://webrtc-review.googlesource.com/c/114340
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26009}
Configuring video decoding and rtp depacketization through json was introduced
in a prior change. This change introduces some basic configurations that will
be used in the initial round of fuzzers that are being added.
TBR=henrik.lundin@webrtc.org
Bug: webrtc:9599
Change-Id: I58aba6a6f24f8374126547deeef0ff4d1708327b
Reviewed-on: https://webrtc-review.googlesource.com/c/113834
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26005}
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}
These metrics by themselves won't be as useful, unless they can be correlated to the use of the
feature 'WebRtcHideLocalIpsWithMdns'. This can be done by running a finch experiment where we turn
the feature on for a % of users, we can then compare these metrics for users with and without
the feature turned on.
A complementary change is required in Chrome:
tools/metrics/histograms/enums.xml
Bug: webrtc:9605 webrtc:10091 chromium:914452
Change-Id: Ibc6d16dec95a8e3943ce40063c02903769fe1cb4
Reviewed-on: https://webrtc-review.googlesource.com/c/113321
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26003}
eglDestroyContext has been observed to deadlock with other GL threads
unless the GL program is detached beforehand.
TBR=sakal
NO_TRY=TRUE
Bug: b/120481228
Change-Id: Ie256e745828997b6fee0d62e681f5ef953aa0fe7
Reviewed-on: https://webrtc-review.googlesource.com/c/114164
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25999}
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}
The address and the related address of local candidates are sanitized
accordingly when the mDNS concealment of local IPs is enabled. Also,
remote hostname candidates created from signaling are sanitized in stats
as well. A couple of unit tests are revised to reflect the desired
behavior of AsyncResolverInterface so that when a hostname candidate is
resolved, the hostname is kept in the candidate address.
Bug: webrtc:9605, chromium:914452
Change-Id: Iad9ad04ce4e50304e44cf04b15b97a7ae2dec960
Reviewed-on: https://webrtc-review.googlesource.com/c/113643
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25996}
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}
It is important that these numbers do not change, so instead of
referring to constants we will use literals here. If we need to update
them we will simply have to update this test as well.
Bug: webrtc:7452
Change-Id: I2808ef08d2236c10666258a8670cc2fd08543143
Reviewed-on: https://webrtc-review.googlesource.com/c/114160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25991}
- 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}
Using declarations should use fully qualified names (with leading `::`)
unless they are referring to a name inside the current namespace.
Source: https://abseil.io/tips/119.
This CL removes a lot of "using webrtc::*" adding a namespace to the
tests. It also removes some unneeded "using" declarations.
Bug: webrtc:9855
Change-Id: Id6eb843e9dcee2e458b1ffd0c499df390fa9c45d
Reviewed-on: https://webrtc-review.googlesource.com/c/114001
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25987}
When migrating the audio device, we accidentally dropped a /2 for
PlayoutDelay. This meant we would estimate a delay of 150ms instead of
75ms for JavaAudioDeviceModules. This change fixes that.
Bug: webrtc:7452
Change-Id: I20b70ebf141410209953243ae665644b92e480f5
Reviewed-on: https://webrtc-review.googlesource.com/c/113946
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25986}
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}
rtc::s_url_decode internally calls transform on rtc::url_decode which operates
on raw char buffers. This is used in some core parts of ice server parsing so
it makes sense to add at least a basic fuzzer here. Corpus generation will be
tailored in a future CL.
Bug: webrtc:10117
Change-Id: If1685601c746c4a9f88c2a8d396eeb3f1b1688d4
Reviewed-on: https://webrtc-review.googlesource.com/c/113835
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25980}
