VideoSimulcastConfig::encoding_params and VideoConfig::min/max_encode_bitrate_bps
were replaced by VideoConfig::encoding_params.
All usage of the previous options has been updated to the new option.
Bug: webrtc:11607
Change-Id: I52cd9efa6e640929485da7aa1e61d15a1a693b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261949
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36865}
By calling release() the ref counter is not decremented and when the
observer is later wrapped in a scoped_refprt its counter will be 2
instead of 1, leading to a memory leak.
Bug: b/231912631
Change-Id: I466e5c29ba4f4aee43e8cf23f2e04fa4b6351b4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261954
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36862}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
* Paramaterize VideoReceiveStream2Test to have variations that run with
and without a metronome.
* Migrate over tests to ensure frame timing is used.
Bug: webrtc:14003
Change-Id: Icccc2f0d548aaa64c50e010056e1e651174e02fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260942
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36812}
* Structs with user-declared constructors are no longer considered
aggregates, so remove the declarations when possible
* Types of both arguments to "==" must match to avoid "ambiguous
function call" warning
* Various types of math involving enums are deprecated, so replace with
constexprs where necessary
* ABSL_CONST_INIT must be used on definition as well as declaration
* volatile memory may no longer be read from and written to by the same
operator, so replace e.g. "n++" with "n = n + 1"
* Replace an outdated check for no_unique_address support with
__has_cpp_attribute
* std::result_of(f(x)) has been removed, replace with
std::invoke_result(f, x)
Bug: chromium:1284275
Change-Id: I77b366ab1da7eb2c1e4c825b2714417c31ee5903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261221
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36786}
The webrtc::VideoStreamDecoderInterface was basically created as a public version of FrameBuffer2, but to hide the complexity of FrameBuffer2 it was also combined with decoding so that the public API could be reasonably simple to use. FrameBuffer3 has a simple API with a clear purpose, so its API can be exposed directly.
Bug: webrtc:14026
Change-Id: I81dc84b869e4d16c5e02feb5c876fbcede3d4a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36781}
Metrics for calculating sent/recv key frame rate:
num_encoded_frames -- number of encoded frames
num_decoded_frames -- number of decoded frames
num_send_key_frames -- number of sent keyframes
num_recv_key_frames -- number of received keyframes
Metrics for frame size (in bytes) distribution:
recv_key_frame_size_bytes -- key frame size
recv_delta_frame_size_bytes -- delta frame size
The stats are only reported if the user so requests. This is
because the stats may not be useful for regression tracking,
but mainly for adhoc analysis.
Tested:
```
~/s/w/src (frame_stats↑1|✚1) $ autoninja -C out/Debug && out/Debug/test_support_unittests --gtest_filter="PeerConnectionE2EQualityTestSmokeTest.Smoke" 2>/dev/null
Note: Google Test filter = PeerConnectionE2EQualityTestSmokeTest.Smoke
[==========] Running 1 test from 1 test suite.
[----------] Global test environment set-up.
[----------] 1 test from PeerConnectionE2EQualityTestSmokeTest
[ RUN ] PeerConnectionE2EQualityTestSmokeTest.Smoke
...
RESULT num_encoded_frames: Smoke/alice-video= 31 count_biggerIsBetter
RESULT num_decoded_frames: Smoke/alice-video= 31 count_biggerIsBetter
RESULT num_send_key_frames: Smoke/alice-video= 2 count_biggerIsBetter
RESULT num_recv_key_frames: Smoke/alice-video= 2 count_biggerIsBetter
RESULT recv_key_frame_size_bytes: Smoke/alice-video= {1847,126} count_biggerIsBetter
RESULT recv_delta_frame_size_bytes: Smoke/alice-video= {1150.1034,393.66329} count_biggerIsBetter
...
RESULT num_encoded_frames: Smoke/charlie-video= 31 count_biggerIsBetter
RESULT num_decoded_frames: Smoke/charlie-video= 31 count_biggerIsBetter
RESULT num_send_key_frames: Smoke/charlie-video= 2 count_biggerIsBetter
RESULT num_recv_key_frames: Smoke/charlie-video= 2 count_biggerIsBetter
RESULT recv_key_frame_size_bytes: Smoke/charlie-video= {1847,126} count_biggerIsBetter
RESULT recv_delta_frame_size_bytes: Smoke/charlie-video= {1150.1034,393.66329} count_biggerIsBetter
...
[----------] 1 test from PeerConnectionE2EQualityTestSmokeTest (2196 ms total)
[----------] Global test environment tear-down
[==========] 1 test from 1 test suite ran. (2196 ms total)
[ PASSED ] 1 test.
```
Bug: webrtc:14019
Change-Id: I3d5ea0cf659a6299c6d94f5004cbf0c763b267e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260924
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36759}
With the encoding parameters in the SimulcastConfig objects, it wasn't
possible to configure explicit encoding parameters when using singlecast,
required for example to use the spec standard SVC API.
Bug: webrtc:11607
Change-Id: I92b1446e772e2ecec93379dc91a3da159b8bc209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260002
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36731}
This is an implementation API, user classes should in principle
only use the channel_interface.h
Bug: webrtc:13931
Change-Id: I85c285217858dc087c90a50792e980f731f4439f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260185
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36674}
This is the first step in the next refactoring:
1. Make general peer params immutable and accessible only via const ref.
2. Extract params which can be changed during the call
3. Expose mutators for mutable params protecting them with proper
locking.
Bug: b/213863770
Change-Id: Ie3ac17918f1fed3b8ec84992f8b0afc402bce7f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260003
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36647}
Return a copy of the frames counters to be able to get DVQA stats while
it is running.
Bug: b/213863770
Change-Id: I04755117cf363e1f160caa11dbe52fb30324fdf5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259867
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36638}
media_kind is the old name (that is kept around since we can't deprecate)
BUG=None
Change-Id: I445441a54bb4ff408502d1aba6834cdac874324b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259766
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36625}
While the target has a restricted visibility, since it was in rtc_base_approved
public deps, a lot of targets were able to bypass the visibility check.
So we remove the visibility restrictions and use the dependency explicitely
everywhere instead.
Bug: webrtc:8603
Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36566}
Now that FieldTrialsView is (almost) a supported object,
let downstream tests use the ScopedKeyValueConfig for
testing.
Bug: webrtc:10335
Change-Id: I7bf7577387f1123b473275990ca32b80a8ddf9bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258480
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36519}
There doesn't seem to be any tests currently running with
VideoToolbox H.264 codec.
Add a minimal test to make sure there are no regressions.
Bug: webrtc:13934
Change-Id: I664833d06e9b4f919b0662781891f5448e6057c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257922
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36511}
The features have two safety fallbacks:
- multichannel config has a killswitch WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch
- stereo detection has a killswitch WebRTC-Aec3StereoContentDetectionKillSwitch
Both features are enabled by default in the AEC3 config.
Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I340cdc9140dacd4ca22d0911eb9f732b6cf8b226
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258129
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36482}
This CL replaces those references with the smallest set of targets
that can satisfy the linker dependencies revealed by building the
"all" target.
Bug: webrtc:13634
Change-Id: Ia778630b18e1164138c41d245c3c8effed67f8e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257282
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36445}
This explores the theory that targets that have no files, just
dependencies, are unnecessary.
Bug: webrtc:13805
Change-Id: I1feb50cf3886128031af8970eae361e35fb052c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256974
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36363}
convert rtc_base/network and collateral.
This also remove last usage of system_wrappers/field_trials
in p2p/...Yay!
Bug: webrtc:10335
Change-Id: Ie8507b1f52bf7f3067e9b4bf8c81a825e4644fda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36357}
This patch takes a stab at modules/video_coding,
but reaches only about half.
Bug: webrtc:10335
Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36335}
This metric weights the PSNRs of luma and chroma planes in
a slightly smarter way than our current PSNR metric.
> J. Ohm, G. J. Sullivan, H. Schwarz, T. K. Tan and T. Wiegand,
> "Comparison of the Coding Efficiency of Video Coding Standards—Including
> High Efficiency Video Coding (HEVC)," in IEEE Transactions on Circuits and
> Systems for Video Technology, vol. 22, no. 12, pp. 1669-1684, Dec. 2012
> doi: 10.1109/TCSVT.2012.2221192.
Bug: None
Change-Id: Iec105e0b491628fc0ad4be9155b991203846ad1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256463
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36311}
This patch fixes a problem for test programs that mix usage of
ScopedKeyValueConfig and the global field trial string.
In this case, tests that were using CallTest.
The solution is to check the global string when nothing has been explicitly added to a ScopedKeyValueConfig.
Bug: webrtc:13828
Change-Id: Ib89735670cfe93340ca0a8bac53f8a64a600ad66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256366
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36294}
convert almost all of video/ (and the collateral)
Bug: webrtc:10335
Change-Id: Ic94e05937f54d11ee8a635b6b66fd146962d9f11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36192}
This cl/
1) move WebRtcKeyValueConfig from api/transport to api/ directory.
2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials).
3) removes usage of webrtc::field_trial:: from the pc/ directory.
4) removes a few unused includes of system_wrappers/field_trial.h.
Bug: webrtc:10335
Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36160}
The problem was fixed by implementing the methid PacketDuration() in
AudioDecoderG722StereoImpl, which catches the issue in
AudioDecoder::Decode().
Bug: chromium:1280851
Change-Id: I31f974b9999f3c1c62b0e5dc39bb3e56a9a9388d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251842
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36034}
The flags isolated-script-test-output and isolated-script-test-perf-output need to be consumed by the tests.
The generated .app folder in added in the data list of the gni file.
This will make it available in the runtime_deps file and thus will be populated to the swarming tasks.
Bug: webrtc:13556
Change-Id: I2c75774b847d9f686c3abc00ba0400bbc3fcefae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240520
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36029}
This emulates behaviour from frame buffer 2, but does not handle stats.
In contrast to frame buffer 2, all work happens on the same task queue.
FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind
a field trial WebRTC-FrameBuffer3.
This separates frame scheduling behaviour into a few components,
VideoReceiveStreamTimeoutTracker
* Handles the stream timeouts.
FrameDecodeScheduler
* Manages the scheduling and cancelling of frames being sent to the
decoder.
FrameDecodeTiming
* Handles the timing and ordering of frames to be decoded.
Other changes
* Adds CurrentSize() method to FrameBuffer3
* Move timing to a separate library
* Does a thread check for Receive statistics as this is now
on the worker thread.
* Adds `FlushImmediate` method to RunLoop so that
video_receive_stream2_unittest can pass when scheduling is happening
on the worker thread.
Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721
Bug: webrtc:13343
Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35847}
This is a test-only class that today has high precision for its frames.
We intend to make the default task queue precision lower, so high
precision use cases have to opt-in to kHigh if they want to continue to
use high precision.
Bug: webrtc:13604
Change-Id: I25babec2a64d91d45548ad017200b806a60efe11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249362
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35808}
This makes fuzzer test cases fail if there's an assert failure in
the helper functions called by the test.
Bug: None
Change-Id: Ic187d72b8d4e016659a68a7bdcaadb78ab2aab05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246400
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35804}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
Use cases of TaskQueue or TaskQueueBase that are considered high
precision are updated to make use of PostDelayedHighPrecisionTask
(see go/postdelayedtask-precision-in-webrtc) instead of PostDelayedTask.
The cases here are the ones covered by that document, plus some
testing-only uses. The FrameBuffer2 and DataTracker use cases will
be covered by separate CLs because FrameBuffer2 uses
RepeatingTaskHandle and DataTracker uses dcsctp::Timer.
This protects these use cases against regressions when PostDelayedTask
gets its precision lowered.
This CL also adds TaskQueue::PostDelayedHighPrecisionTask which calls
TaskQueueBase::PostDelayedHighPrecisionTask (same pattern as for
PostDelayedTask).
Bug: webrtc:13604
Change-Id: I7dcab59cbe4d274d27b734ceb4fc06daa12ffd0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248864
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35781}
Because rtc::Thread inherits from TaskQueueBase, it already implements
a pair of PostTask/PostDelayedTask methods that we want to keep. But in
addition to those, rtc::Thread defines its own PostTask/PostDelayedTask
using templates. These are the versions that we want to deprecate.
They were originally implemented prior to rtc::Thread inheriting from
TaskQueueBase. We want to deprecate them because...
- We don't want to have multiple code paths that do the same thing.
- We want to move away from rtc::Thread to TaskQueueBase long-term.
- These versions are not overridable in Chromium.
- These versions don't have high/low precision versions of PDT.
Helper methods are added to rtc::Thread so that callers don't have to
wrap every lambda in webrtc::ToQueuedTask() and update dependencies.
Bug: webrtc:13582
Change-Id: I58702c53f4cb3705681bd9f1ea16b7aaa5052c18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35750}
Feedback can include lost packets. Ensure FeedbackGenerator support lost packets to be able to use this class properly when testing with loss.
Bug: none
Change-Id: Ibd740dfae358c0543fbee62cd40ef13a7ac1123f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247372
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35740}
Updates all webrtc code, to have a small followup cl to just add the
"explicit" keyword. Patchset #24 passed all webrtc tests, with explicit.
Bug: webrtc:13464
Change-Id: I39863d3752f73209b531120f66916dc9177bf63a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242363
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35718}
ChangeResolution
Instead a warning is logged.
This effect tests only, and allows us to write screen share tests that may try to trigger the capturers to change resolution.
Bug: none
Change-Id: I4740fc4ed0bcf75e1c9df332fa610c24ed14973a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245981
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35668}
In order to unify WebRTC recipes with Chromium recipes this CL tries to revert the old CL https://webrtc-review.googlesource.com/c/src/+/171681.
This CL was already partially reverted (https://webrtc-review.googlesource.com/c/src/+/171809).
In upcoming CLs, the added flag dump_json_test_results will be removed in order to use isolated-script-test-output instead.
Bug: webrtc:13556
Change-Id: I3144498b9a5cbaa56c23b3b8adbac2229ad63c37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245602
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35666}
Add TODO for accessing `previous_demuxer_criteria_`, currently accessed
from two threads (unsafe).
Changed RtpDemuxerCriteria to be a class, all members private with
accessor methods instead of direct variable access. Moving forward
this can allow for things like checking for thread/sequence and state
consistency.
Bug: webrtc:12517, webrtc:11993
Change-Id: I21c1b3067e988494ce6f4c6c85c62165801883bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244083
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35616}
Important: This change does not in any way affect echo cancellation or standardized stats. The user audio experience is unchanged. Only non-standard stats are affected. Echo return loss metrics are unchanged. Residual echo likelihood {recent max} will no longer be computed by default.
Important: The echo detector is no longer enabled by default.
API change, PSA: https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ
This CL removes the default usage of the residual echo detector in APM.
It can now only be used via injection and the helper function webrtc::CreateEchoDetector. See how the function audio_processing_unittest.cc:CreateApm() changed, for an example.
The echo detector implementation is marked poisonous, to avoid accidental dependencies.
Some cleanup is done:
- EchoDetector::PackRenderAudioBuffer is declared in one target but is defined in another target. It is not necessary to keep in the API. It is made an implementation detail, and the echo detector input is documented in the API.
- The internal state of APM is large and difficult to track. Submodule pointers that are set permanently on construction are now appropriately marked const.
Tested:
- existing + new unit tests
- audioproc_f is bitexact on a large number of aecdumps
Bug: webrtc:11539
Change-Id: I00cc2ee112fedb06451a533409311605220064d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239652
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35550}
FrameBuffer3 keep track of order, decodability and continuity of the inserted frames. Compared to FrameBuffer2 which schedule frames for decoding and is thread safe, FrameBuffer3 does not schedule decoding and is thread unsafe.
Change-Id: Ic3bd540c4f69cec26fce53a40425f3bcd9afe085
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238985
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35494}
This was a remenant leftover from a previous design, which was no longer
valid after the switch to TaskQueues. ReturnReason::kStopped was not
used at all, and so Timeout or FrameFound can be inferred from whether
the frame is null or not.
Bug: webrtc:13343, webrtc:13346
Change-Id: Ib0f847b1e1192e32ea11208e48f5a3892703521e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239651
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35490}
Sender isn't actually require to identify the stream, so specifying it
every time is useless. This CL removes sender from StatsKey object and
introduces StreamsInfo object which contains all required metadata about
streams that are seen by DVQA.
Bug: b/205824594
Change-Id: I5b6be3865a30fd5980ff6e7e50906abe70a632ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238562
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35399}
It will ensure that clock moved forward, so clock related metrics as
CPU usage will be also calculated correctly. We should dig into it deeper
to fix the root cause later.
Example failure:
https://ci.chromium.org/ui/p/webrtc/builders/try/win_asan/44610/overview
Bug: b/205824594
Change-Id: If1ebcf6a2b88ba0054479be292cca8f50506e741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238381
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35379}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
Only used in unit tests and a duplication of what `capture_output_rms_`
already does.
This CL also removes `AudioProcessingStats::output_rms_dbfs`, which is
now unused.
Bug: webrtc:5298
Fix: chromium:1261339
Change-Id: I6e583c11d4abb58444c440509a8495a7f5ebc589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235664
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35246}
Also stop using ApplyConfig() and in [1] fix the build errors when
WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE is defined.
[1] modules/audio_processing/test/audio_processing_builder_for_testing.cc
Bug: webrtc:5298
Change-Id: I50dc5668b952e7ca7fa83c7a5182c013e928c450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235365
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35228}
DefaultVideoQualityAnalyzerFramesComparator::Stop() may not block until
all frames comparisons are processed in case when new comparison was
added after worker thread checked for available comparisons and Stop()
was invoked before worker thread checked the state_.
Bug: webrtc:13277, webrtc:13240
Change-Id: Ic16fdc01e43c04529cd83e5d9ef66d7573973cfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235205
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35212}
A preparation for splitting server sockets out into a separate
interface, see https://webrtc-review.googlesource.com/c/src/+/232607.
Transition plan:
1. Land this cl.
2. Update downstream code to use the new name.
3. Attempt landing
https://webrtc-review.googlesource.com/c/src/+/232607. May need
additional steps to not break downstream implementations of
PacketSocketFactory::CreateServerTcpSocket.
Bug: webrtc:13065
Change-Id: Ife448c705222f4c9f66a096e3dc7eb07e0f9c3af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35155}
Use config from FakeEncoder in some tests.
Bug: none
Change-Id: I1d7e01f604f8aabb5d6815bb519ef2532d024d76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233243
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35138}
Unlike ReadBits, ConsumeBits doesn't limit number of bits it may advance,
and thus should work when that number is close to the integer limit
Bug: chromium:1250730
Change-Id: Ia7847869ef9d3fc16450d572c9e2be6e1aa36741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232332
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35042}
Thanks to the elimination of `ExperimentalNs`, there is no need anymore
to pass `webrtc::Config` to build APM.
Hence, `AudioProcessingBuilder::Create(const webrtc::Config&)` is also
removed.
Bug: webrtc:5298
Change-Id: I0a3482376a7753434486fe564681f7b9f83939c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232128
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35025}
This is tested by a simple unit test and a new fuzzer that verify that all that can be parsed also can be written.
Bug: webrtc:12000
Change-Id: I461aedf97d3dec6e8916e72110fa097c3b31c27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34986}
The VP9 encoder may drop a frame internally which will not advance the
frame pattern. Consider the following scenario where only spatial layer
0 and temporal layer 0 is active:
1. Key frame encoded
2. Spatial layer 1 is activated
3. Delta T0 dropped
4. Delta T0 encoded
No S1T0 frame is encoded in (1) since it's not active. When
NextFrameConfig is called in (3) it will say that future frames may
reference T0 on both S0 and S1, but it's then dropped.
On step (4), the SVC controller essentially thinks it's encoding a new
picture and will happily reference the T0 on what it thinks is the first
delta frame. However, this is actually still the key frame and since
there was no S1T0 frame produced it will reference an invalid buffer.
To fix this, only say it's possible to reference a T0 frame after it has
been successfully encoded.
Bug: webrtc:11999, webrtc:13142, chromium:1178444
Change-Id: Iab3d2042ce0b3fa7d952b2831d1a36b1a6613a86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231695
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34982}
With the new SVC controller this will hopefully help uncover more subtle
bugs.
Bug: webrtc:11999
Change-Id: Iab76d38b3fb8dfbbeb269f4ba1e74f6f425501f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231694
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34981}
So that applications don't need to construct it from the exposed
network_thread.
The EmulatedNetworkManagerInterface::network_thread() accessor is currently
used as a way to get to emulation's SocketServer, and should be deleted
when applications of the emulation framework have migrated away from
that usage.
Bug: webrtc:13145
Change-Id: I3efa55d117cad8ac601c48a9d2d2aa62a121f9c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231649
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34964}
