and ensure there is only one, similar to what is done with RTX.
This avoids exposing a payload type there.
See also
https://github.com/w3c/webrtc-pc/issues/2696
BUG=webrtc:42221750,webrtc:360058654
Change-Id: Id7c2ddeaf47a3169db9be43c9c5b8e59346f1d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376760
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43929}
This CL fixes two issues with the old way targetBitrate was reported:
1. The target is per encoder, i.e. per SSRC, but the old way to report
it was per sender and was approximately the sum of all encodings'
targetBitrate in most cases.
2. The old value did not come directly from the VideoBitrateAllocation
and tended to be greater than the sum of all targets (don't know
why).
We know the old value was wrong and the new value correct because
the actual bytes produced by the encoder closely matches the configured
target, which wasn't always the case with the old metric implementation.
Tested with unit tests and manually in Chrome by going to
https://henbos.github.io/codec-quality/src/index.html and ensuring
target ~= actual bytes produced. It also matches the debug logging of
video_stream_encoder.cc.
Bug: webrtc:42225524, chromium:392424845
Change-Id: I7a6f69e053ebc3fd972c2c4b7712750e721c0acc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376460
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43854}
and make it less opus-specific.
BUG=None
Change-Id: I6fe2975ba6e45a3758fedc5b950de90e8d9df362
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375436
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43846}
For-loop has a 'continue' statement that skips increment of the index.
Added such an increment before 'continue' for the index to keep up with
the for-loop.
I guess current implementation will bug on codecs sequence like:
'red, unknown, opus'
since the subsequent for-loop (the 'red_codec' one) will not be able to
find 'opus'.
Seems like adding second increment statement is the easiest way to fix it.
Bug: None
Change-Id: Iab9cc66cf569458af9fd9ba5b938d83186c78c73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43488}
also fix a few TODOs
Bug: None
Change-Id: I2d287ed1a58f71ef32d5dc5624879ae8c63348b5
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370122
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43485}
The RTCP mode is a send property for both send and receive channels. Send properties should be configured based on what peers support/prefer, which is described by the remote description (content).
Bug: webrtc:340041654
Change-Id: I18cd59e98aecfbbd8f4919b98381836184c10d77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368980
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#43449}
This seems to have no effect on tests, so it appears that these were
not used after all.
The goal is to make transport-cc a media-section-level attribute.
Bug: webrtc:378698658
Change-Id: Ia20ca5b91472b02db30f911ad1a1892cf36cd682
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43411}
We should use the Timestamp type, rather then int64, to store timestamps. In https://webrtc-review.googlesource.com/c/src/+/365001/ an additional int64 timestamp was added (last_sender_report_timestamp_ms).
This CL fixes the new timestamp, as well as other similar timestamps in MediaReceiverInfo (last_sender_report_utc_timestamp_ms and last_sender_report_remote_utc_timestamp_ms).
Bug: webrtc:372393493
Change-Id: I0e473730e85a69ec595b421e2c3db920364008eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43248}
Add an environment clock timestamp to SenderReportStats and make it visible in rtc_stats_collector.cc. This make it possible to use the pc->GetConfiguration().stats_timestamp_with_environment_clock() flag to decide which timestamp to use when creating a RTCRemoteOutboundRtpStreamStats object.
This CL is the third (and possible the last) of a series of CLs that aim to replace the UTC timestamps in RTCStats objects to Environment clock timestamps. The other CLs where https://webrtc-review.googlesource.com/c/src/+/363946 and https://webrtc-review.googlesource.com/c/src/+/364782.
When Chromium and Google internal uses of RTCStats are updated to set the stats_timestamp_with_environment_clock configuration, the flag can be deleted.
Bug: chromium:369369568
Change-Id: Ic0b07d7b012505267bd6516f19a9ba90df4cafab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365001
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43206}
I missed one timestamp in https://webrtc-review.googlesource.com/c/src/+/363946, meaning that the config flag that was added do not yet work for all timestamps in RTCStats objects. The RTCRemoteOutboundRtpStreamStats still has UTC timestamps even if the config flag is set.
I will solve this by saving both an UTC (existing) and env (to be added) timestamp, and then let rtc_stats_collector choose timestamp based on the value of the config flag (just like RTCRemoteInboundRtpStreamStats is done in the 363946 commit).
Before adding the new env_ timestamp I want to make this change. I rename the existing timestamp to show what epoch it uses (NTP or UTC). This will later make it clear which timestamp is which.
So this CL will make no logical change, just renaming members.
I only need to rename the last_sender_report_timestamp_ms, but opted to rename the remote timestamp as well, to be consistent with the naming convention I add in this CL.
Bug: chromium:369369568
Change-Id: Icfe7cf274995b39799e1478a1bb8cdf5134f0b16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364782
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43194}
Note: Does not include code for the actual late allocation
of PTs.
Bug: webrtc:360058654
Change-Id: Iaa6bd2db2f974aad84fe1ae9c1aca5aea5d1d25e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362320
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43014}
This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code.
Bug: webrtc:360058654
Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42989}
Will be later used to conditionally enable mixed codec simulcast
with a field trial.
Bug: webrtc:42220378
Change-Id: I527a488c04cd2b5a9f4ec703504b67943e966ab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361403
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42929}
reduced-size RTCP, i.e. not prefixing RTCP packets with either a sender report or receiver report has been implemented for a long time but only for video.
This CL adds it for audio as well. This reduces the size of audio NACKs (16 bytes, typically one NACK per packet) sent by not prefixing it with a receiver report (32 bytes).
Other packets are not affected as e.g. transport-cc feedback does not add a RR even though that is technically required.
The effect on NACK can be tested by running Chromium with
--disable-webrtc-encryption --force-fieldtrials=WebRTC-FakeNetworkReceiveConfig/loss_percent:5/
against this fiddle negotiating audio nack:
https://jsfiddle.net/fippo/8ubtLnfx/1/
BUG=webrtc:340041654
Change-Id: I06fb94742ff1b6f9a464c404bfc53913f23498d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350269
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42330}
These are aliases for cricket::Codec.
Also remove internal usage
Bug: b/42225532
Change-Id: I220b95260dc942368cb6280432a058159eec8700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42194}
by creating an id collision and letting UsedIds resolve it.
Also avoid id=15 which is forbidden by
https://www.rfc-editor.org/rfc/rfc8285#section-4.2
so might cause interop issues in offers to implementations
not supporting two-byte extensions.
BUG=webrtc:15378
Change-Id: I27926f065f8e396257294da7acf2be9802169805
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41696}
instead of throwing an error when trying to pick a send codec.
BUG=webrtc:15145,webrtc:4957
Change-Id: I056b145c093348576e1aeaf5def50d5414f2de70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330122
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41360}
Because of our asymmetrical codec situation, it's possible to have
send only codecs that we cannot negotiate even with ourselves.
This means that we should not have a DCHECK, but just a plain error.
Bug: webrtc:15064
Change-Id: I0c170e5c7f356197bcb04bcecb8259c344423ccb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323183
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40939}
This makes it consistent with how things are done in webrtc_video_engine.cc
This will improve the JS code by not having to initialize an audio
track every time frames need to be sent over, especially from another
peer connection in case of encoded transforms.
Bug: chromium:1477192
Change-Id: I3f938ad812ff377599a3799d4c2d2cd85149189e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322702
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40917}
following the previous change to rename the classes derived from
cricket::RtpParameters
Also rename ChangedRecvParameters to ChangedReceiveParameters.
BUG=webrtc:13931
Change-Id: Ia51dd39905a5cbb98162c3948930e43ccaf3786d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314500
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@{#40677}
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.
Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
Media*Channel objects used to subclass webrtc::Transport.
This was not an optimal design. This CL makes the transport
a member variable of MediaChannelUtil.
Bug: None
Change-Id: I85d33cc1b32b931e563b7bb2d277f1c512600831
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309800
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40328}
Added as stopped by default as it should be requested by the application,
but it should be listed as available.
Bug: webrtc:14631
Change-Id: I301cfd29c79083c97b4a43b8fdafee2dbe4887a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308824
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40300}
into separate sections for each implemented class.
Bug: webrtc:13931
Change-Id: I600f49f3fb195761d13d304f112f36c7c62689df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308120
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40260}
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
This ensures that the MediaChannel interface is only implemented
through a send/receive shim, splitting channels also when kBoth is
used.
Bug: webrtc:13931
Change-Id: Ie97809597eaae7b4f504939339795432c34e56cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307461
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40210}
Add an API to pass AudioFrameProcessor as a unique_ptr.
Bug: webrtc:15111
Change-Id: I4cefa35399c05c6e81c496e0b0387b95809bd8f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301984
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40187}
And fix bug that prevented it from passing.
Bug: webrtc:13931
Change-Id: I6cbc8e3aad704f6f7e33362efb7ec589ca6e6568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306184
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40112}
This makes the handling somewhat more uniform, and is the same
for both video and audio channels.
Bug: webrtc:13931
Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40107}
