This adds a Call-based test, that sets up video-pipeline with a VP8
encoder and the corruption detection header extension configured.
It then verifies that the corruption likelihood metrics are populated
in the receive stream stats.
Bug: webrtc:358039777
Change-Id: Ide005459a801778de4238e786f13efc8c3245f3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365860
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43254}
This requires making CallConfig move-only so it can hold a unique_ptr to
the factory, but as discussed with Danil, that seems fine.
Bug: chromium:355610792
Change-Id: Ie52e33faaa4a2af748daeb25f5327b7a532936e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42679}
* Simplified ctor. Get settings (max_qp, content_type, etc) from encoder_config passed to CreateEncoderStreams().
* Some tests assigned VideoEncoderConfig::video_stream_factory to EncoderStreamFactory they created. That's not really needed. VideoStreamEncoder creates the factory if video_stream_factory is not provided [1]. Removed video_stream_factory initialization in tests.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;l=1002;drc=1d7d0e6e2c5002815853be251ce43fe88779ac85
Bug: b/347150850, webrtc:42233936
Change-Id: Ie0322abb6c48e1a9bd10e9ed3879e3ed484fea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355321
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42608}
Replace usage of BuiltInNetworkBehaviorConfig.link_capacity_kbps in tests with DataRate link_capacity.
Bug: webrtc:14525
Change-Id: Id1c1b8d20eb2be5e9d1461704bb7c37c61c491f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350300
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42306}
Instead of from the global field trial string.
Bug: webrtc:42220378
Change-Id: Iddb41429e388792de02f702b4caa35689c57d9e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347720
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42201}
Old target and call/simulated.h exist but refer to new target in test/network.
Bug: webrtc:14525
Change-Id: Ida04cef17913f2f829d7e925ae454dc40d5e8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349264
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42191}
Instead of passing it as optional parameter during construction, pass field trials as required parameters on use.
Test that create the EncoderStreamFactory might not have an easy access to the actual field trials, but prod code has appropriate field trials when uses the factory.
This way EncoderStreamFactory doesn't need to depend on global field trial string through FieldTrialBaseConfig class.
Bug: webrtc:10335
Change-Id: I8f7030e41579ff2c5dd362c491a4e1624b23e690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42098}
Some of the fake encoders, FakeVp8Encoder in particular, reuse structures that in turn rely on field trials. Thus fake encoders also can benefit from Environment passed at construction.
Bug: webrtc:15860
Change-Id: Ia1542b2663c75fd467e346aad9ead627ff9b3b0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346780
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42046}
describing video codecs with their parameters as static members of SdpVideoFormat:
static const SdpVideoFormat VP8();
static const SdpVideoFormat H264();
static const SdpVideoFormat VP9Profile0();
static const SdpVideoFormat VP9Profile1();
static const SdpVideoFormat VP9Profile2();
static const SdpVideoFormat VP9Profile3();
static const SdpVideoFormat AV1Profile0();
static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.
BUG=webrtc:15703
Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41833}
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.
Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
To reduce number calls to the CreateVideoDecoder
Bug: webrtc:15791
Change-Id: I5d6ecc2e5e68165d4e012b3ad7edb6eaa40e1913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336420
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41706}
This CL adds [[deprecated]] to the old signatures, and uses the new
signatures throughout.
Bug: webrtc:14870
Change-Id: Ic9a8198ac0a2f954e1b2e7d05a55dbe04342f958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314962
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40517}
This extension is documented to carry one bit: Screenshare.
It's been used for carrying simulcast layers and experiment IDs.
This CL removes that usage.
Bug: webrtc:15383
Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40457}
It is partial reland, which adds call to Start() to all relevant places,
but doesn't actually switches frame generator to not produce frames from
the moment it was created.
Bug: b/272350185
Change-Id: I6e3bd7af6f5cd8d9baff79c2aada7b2ddfae1c8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310782
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40379}
Most of the usage of the H264Encoder::Create(codec) method passes a
simple codec with just the H264 codec name. This simplified the call
sites in many places and removes references to the codec types.
Bug: webrtc:15214
Change-Id: I4039c0be4ce6e3147c14c7853df4635f344b7d70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307222
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40214}
Prior to this CL, the video `jitterBufferDelay` stat was the accumulated current delay, which is a smoothened version of the target delay. This is not correct according to the spec [1]. Rather, the stat should be the accumulated time spent in the jitter buffer, for all emitted frames. This CL fixes this spec compliance problem.
Expect changes to test metrics and product monitoring as this CL rolls out.
[1]: https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay
Tested:
1. Go to https://jsfiddle.net/jib1/0L6duga2/show
2. Apply 2.0 seconds of video delay.
3. Notice that "Video jitter buffer delay" is slightly less than 1990ms. (2000ms playoutdelayhint - 10ms render delay - Xms decode delay).
Bug: webrtc:15085
Change-Id: I42805faafd7dd3bcdcf3ad08e751e08d6de38906
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304521
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40138}
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.
Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63.
Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200
Original change's description:
> Delete PacketReceiver::DeliverPacket from all implementations
>
> And fix tests that still depend on extensions to be known by the receiver.
>
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
>
> Bug: webrtc:7135,webrtc:14795
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39184}
Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: I9d03f4952938d176ffee110a707acadc1846457c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39189}
And fix tests that still depend on extensions to be known by the receiver.
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Bug: webrtc:7135,webrtc:14795
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39184}
This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae.
Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104
Original change's description:
> Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
>
> This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425.
>
> Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio
>
>
> Original change's description:
> > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp
> >
> > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
> > Therefore DirectTransport is provided with the extension mapping.
> >
> > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
> >
> >
> > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
> > Bug: webrtc:7135, webrtc:14795
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
> > Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39137}
>
> Bug: webrtc:7135, webrtc:14795, webrtc:14833
> Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Auto-Submit: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#39146}
Bug: webrtc:7135, webrtc:14795, webrtc:14833
Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39157}
This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425.
Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio
Original change's description:
> Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp
>
> PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
> Therefore DirectTransport is provided with the extension mapping.
>
> CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
>
>
> Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
> Bug: webrtc:7135, webrtc:14795
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39137}
Bug: webrtc:7135, webrtc:14795, webrtc:14833
Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220
Owners-Override: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39146}
PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
Therefore DirectTransport is provided with the extension mapping.
CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
Bug: webrtc:7135, webrtc:14795
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39137}
