which come from the a=fmtp:<pt> lines in the SDP and were used as either
std::map<std::string, std:string>
with three aliases,
cricket::CodecParameterMap
SdpAudioFormat::Parameters
SdpVideoFormat::Parameters
Use webrtc::CodecParameterMap in all places.
BUG=None
Change-Id: If47692bde7347834c349c6539b43309d8770e67b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41375}
Replace CallFactory class with a factory function
Bug: webrtc:15574
Change-Id: Ib1d8cff8d7550da3af01693a7bc117a7bd342258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330000
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41321}
PacingController per default use a burst interval of 40ms. The behaviour can still be overriden by using the method SetSendBurstInterval.
Bug: chromium:1354491
Change-Id: Ie3513109e88e9832dff47380c482ed6d943a2f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311102
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41254}
and unify algorithms a bit more.
BUG=webrtc:15214
Change-Id: Ie9903f3e56d25b1dc026367e8ae6817275faa07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328442
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41244}
This will be needed to make some changes to `fuzzer_test`
(crrev.com/c/5053732), but also makes sense generally since those are
only meant to exist in tests, hence for testing.
Bug: chromium:1504840
Change-Id: Ic40178c96753179f485c90abb958874320192a3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328440
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Paul Semel <paulsemel@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41231}
StunServer is updated to ensure registring for receiving packet from the socket is happening on the same thread as where the packets are recevied.
Bug: webrtc:15368, webrtc:11943
Change-Id: I94cc3a47278d5489de7f170c8d43015d1551c437
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328120
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41219}
This is a reland of commit 496893e89e5bc8139e50befcb1a26eadbd829b0d
Original change's description:
> Added an encode/decode test parameterizable via command line
>
> This enables testing different settings without updating code and rebuilding the test binary. Example of command:
>
> video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
>
> Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
>
> Bug: webrtc:14852
> Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41179}
Bug: webrtc:14852
Change-Id: Iccb9af8bf6a6c37704bc58b6e57238b55761b079
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327781
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41194}
This reverts commit 496893e89e5bc8139e50befcb1a26eadbd829b0d.
Reason for revert: Breaks https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/16103/overview
Original change's description:
> Added an encode/decode test parameterizable via command line
>
> This enables testing different settings without updating code and rebuilding the test binary. Example of command:
>
> video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
>
> Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
>
> Bug: webrtc:14852
> Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41179}
Bug: webrtc:14852
Change-Id: Ifdce738058c3e3fc7c76886939add2813405cae7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327722
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41183}
This enables testing different settings without updating code and rebuilding the test binary. Example of command:
video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
Bug: webrtc:14852
Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41179}
* Pass codec factories to the video codec tester instead of creating and wrapping codecs into a tester-specific wrappers in video_codec_test.cc. The motivation for this change is to simplify the tests by moving complexity to the tester.
* Merge codec stats and analysis into the tester and move the tester. The merge fixes circular deps issues. Modularization is not strictly needed for testing framework like the video codec tester. It is still possible to unit test underlaying modules with rather small overhead.
* Move the video codec tester from api/ to test/. test/ is accessible from outside of WebRTC which enables reusing the tester in downstream projects.
Test output ~matches before and after this refactoring. There is a small difference that is caused by changes in qpMax: 63 -> 56 (kDefaultVideoMaxQpVpx). 56 is what WebRTC uses by default for VPx/AV1 encoders.
Bug: webrtc:14852
Change-Id: I762707b7144fcff870119ad741ebe7091ea109ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41144}
std::is_pod is deprecated since C++20. Replace with std::trivial and
std::is_standard_layout. Avoids a lot of warnings.
Bug: chromium:957519
Change-Id: Idb4bde7401c14c0896a84c357ec668b9916f613e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325484
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41117}
To replace CreateTimeControllerBasedCallFactory
Update webrtc tests to use this new function
Bug: webrtc:15574
Change-Id: I2b74cd930ecc4f72dd1e7aa853764ca298b66ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325527
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41076}
Update most of the webrtc tests to use EnableMediaWithDefaults instead of SetMediaEngineDefaults
Bug: webrtc:15574
Change-Id: I489a09e4ea3479dc26829ee0c1235e67bcbca7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325485
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41059}
Reasons:
- the code is no longer used in Chrome
- it is conceptually weird for WebRTC to have JSON parsing in its API
- there are concerns around the reliability of the underlying JSON library
Additionally, this CL removes the rtc_json "poisonous" attribute: the scheme is incompatible and redundant with testonly.
Bug: webrtc:1493351
Change-Id: I0b621b0e3f183df7315919d9c89242fbe387928f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325062
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41014}
kDefaultQpMax=56 was defined in multiple places. Move it to media_constants and split it into two: VPx/AV1 and H26x values. H26x value is set to 51 which is the max bitstream QP value for H264/5.
This CL is expected to be a no-op because:
1. VideoCodec::qpMax value has not changed for VP8/9 and AV1.
2. VideoCodec::qpMax is currently not used by OpenH264 wrapper (wiring it up is out-of-scope of this CL).
3. Previous default qpMax=56 exceeded the max value for H26x (=51). External HW H26x encoders likely clamped it and used 51.
Bug: webrtc:14852
Change-Id: I1d795e695dac5c78e86ed829b24281e61066f668
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324282
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40997}
This field trial is configured via command line flag, so may use flag system directly, reducing dependency on global field trial string.
Bug: webrtc:7101, webrtc:10335
Change-Id: I1e48e0e3fdc251b73a375c6d7f1a46fa4f8a179b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322624
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40897}
This change replaces type of absolute_capture_timestamp_ms_ in
TransformableOutgoingAudioFrame from int to optional uint and makes
the function AbsoluteCaptureTimestamp() inside
TransformableAudioFrameInterface pure virtual.
Bug: webrtc:14949
Change-Id: Id3bdbcba63a5f91105ab198208e4f2b11eb3c7db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319000
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40814}
To avoid name collision with Timestamp type,
To avoid confusion with capture time represented as Timestamp
Bug: webrtc:9378
Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40796}
The synthetic delay added in ZeroHzAdapterMode::OnFrame does not
account for delay with respect to the initial frame post from
FrameCadenceAdapter::OnFrame. Fix this to account for time spent
in contention on the encode sequence.
Bug: webrtc:15456
Change-Id: I63446e8dfe8f62b09d972434a705e912f8a73d69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318420
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40675}
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.
Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
ChannelSendFrameTransformerDelegate::SendFrame() currently only
supports sending frames in a single direction. With this change, we
allow sending received audio frames.
Bug: chromium:1464847
Change-Id: I8113a3278dfce7b2ba709afecc672bc9af9c4a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316600
Reviewed-by: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40643}
When built for chromium, some webrtc implementations are overridden and
are implemented by chrome's "//base". For instance webrtc::Location is
implemented by base::Location. So far so good, the affected targets are
correctly defined in GN to depend on base.
The problem: Most targets in webrtc do not declare correctly their
public_deps. When a public header of a target includes one from its
dependency, the dependency must be a public_deps. The public_deps
instruct GN to forward the capability to use code from the dependency
toward the dependent.
Unfortunately, it is not possible to fix the `public_deps` in webrtc,
because its is disallowed via a presubmit. See:
https://webrtc-review.googlesource.com/c/src/+/30262
WebRTC developers decided not to use `public_deps`, because GN config
are "translated" toward different kind of downstream build system who do
not really support the `public` dependencies concept. Instead WebRTC is
using some "common" configuration applied to all of its targets.
This patch add `rtc_common_public_deps` argument, to let embedders
add the dependencies WebRTC depends on.
Bug: chromium:1467773
Change-Id: I7de43372414a09886fcb07905451e6339c8ecc64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316660
Commit-Queue: Arthur Sonzogni <arthursonzogni@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40595}
This is part of effort to enable HEVC for WebRTC. Parser added here to support parsing of QP/picture size for bitstream, for dynamic adaptation and stream metadata abstraction.
Bug: webrtc:13485
Change-Id: I2fbdf210e72e77989ca87ce285da174df5bedd5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298421
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40572}
The following can lead to ODR violations with symbols present in the
app and in the test module:
gn path out/Perf //:webrtc_perf_tests_module //sdk:helpers_objc
//:webrtc_perf_tests_module --[public]-->
//:webrtc_perf_tests_module_loadable_module --[private]-->
//test:google_test_runner_objc --[private]-->
//test:test_support_objc --[private]-->
//sdk:helpers_objc
After this CL:
gn path out/Debug/ //:webrtc_perf_tests_module //sdk:helpers_objc
No non-data paths found between these two targets.
Bug: b/292472934
Change-Id: If8a6ecab9b34bea0f52fe91b3404d1afeca685fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313520
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40481}
This CL does 2 things:
- Change the DCHECK for payload_type_frequency to a CHECK (so that
this error will be a crash not a divide-by-zero)
- Change the replay helper that was used by the fuzzer to set the
frequency of the packets to the video value (90K).
Bug: chromium:1466826
Change-Id: I39941f250b1782b36a3bcddfd347a016591466ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312700
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40468}
