10 Commits

Author SHA1 Message Date
Harald Alvestrand
a6544377bc Remove not-needed webrtc:: prefixes in pc/
This test drives the new tools_webrtc/remove_extra_namespace.py tool.

Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
2023-11-13 13:23:04 +00:00
Tomas Lundqvist
a26d6ed26f Makes sure that RED is not added twice to the list of codecs when it is used with Opus.
Bug: webrtc:15606
Change-Id: I3ab3ee287f5d2e3a0a46520608e5c0931e0bff90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325180
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#41028}
2023-10-27 15:00:55 +00:00
Florent Castelli
1adea9806d Return error when requested codec is preferred but not negotiated
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}
2023-10-16 13:59:13 +00:00
Philipp Hancke
13b5eb7c47 stats: ensure rtx ssrc is associated with primary ssrc
BUG=webrtc:15529

Change-Id: I3623eede7fc7890677516d78f3ef7a89a287eb8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322221
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40873}
2023-10-05 12:33:34 +00:00
Florent Castelli
43a5dd86c2 Implement codec selection api
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}
2023-08-24 13:18:04 +00:00
Henrik Boström
2fec64484f Fix L1Tx target bitrate bug when the standard API is used.
There are now multiple ways to configure VP9 L1Tx:
- Legacy API: configure legacy SVC and disable encodings, this gets
  interpreted as disabling spatial layers (non-standard API hack).
- Standard API: configure scalability_mode. This can be done either
  with a single encoding or multiple encodings. As long as only one
  encoding is active we get a single L1Tx ssrc, same as legacy API.

Due to a bug, the ApplySpatialLayerBitrateLimits() logic which tweaks
bitrates was only applied in the legacy API code path, not the standard
API code path, despite both code paths configuring L1Tx.

The issue is that IsSimulcastOrMultipleSpatialLayers() was checking if
`number_of_streams == 1`. This is true in legacy code path but not
standard code path. The fix is to look at
`numberOfSimulcastStreams == 1` instead, which is set to the correct
value regardless of code path used.

This CL adds comments documenting the difference between
`number_of_streams` and `numberOfSimulcastStreams` to reduce the risk
of more mistakes like this in the future.

Bug: chromium:1455039, b:279161263
Change-Id: I69789b68cc5d45ef1b3becd310687c8dec8e7c87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308722
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40287}
2023-06-15 12:48:48 +00:00
Henrik Boström
eb99300bb5 Parameterize test that all layers can be inactive.
Previously this test only ran on VP9, now it runs for all codecs.

Bug: webrtc:15080
Change-Id: Id61a261cef3463a22062e3d313dc2725e051773d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300861
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39812}
2023-04-11 15:03:30 +00:00
Henrik Boström
fb65d23d73 Parameterize PeerConnectionEncodingsIntegrationTest for standard paths.
This CL introduces PeerConnectionEncodingsIntegrationParameterizedTest,
which is based on PeerConnectionEncodingsIntegrationTest but covers all
codecs using INSTANTIATE_TEST_SUITE_P (VP8, VP9, H264, AV1).

This applies to all standard paths, which in the case of VP9 and AV1
requires opting in to it by specifying scalabilityMode and
scaleResolutionDownBy. They are also limited to L1Tx because the other
codecs don't support SVC.

The VP9-only tests continue to run as TEST_F with
PeerConnectionEncodingsIntegrationTest.

Bug: webrtc:15079
Change-Id: I3429c90f2f79ff60adad0b33975bccdda31ce6d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300900
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39808}
2023-04-11 14:06:42 +00:00
Henrik Boström
b515c17dba Add SVC tests that all layers can be inactive.
This CL contains [1] and [2], tests that have previously been reverted
due to an issue with VP9 that was fixed in [3]. The re-landed tests
have been renamed as this test suite is now called
PeerConnectionEncodingsIntegrationTest.

[1] https://webrtc-review.googlesource.com/c/src/+/299146
[2] https://webrtc-review.googlesource.com/c/src/+/299008
[3] d26fc16a1b

Bug: webrtc:15033
Change-Id: I4c2b2c0ff3e708ec3a50d38a92214ca9c9ddd8c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300840
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39802}
2023-04-11 08:41:57 +00:00
Henrik Boström
da9e284308 Rename simulcast flow tests: PeerConnectionEncodingsIntegrationTest.
This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests.
The reason for renaming is to reflect that a) this is an integration
test, not a unit test, and b) not all of the tests use simulcast (some
use a single encoding, i.e. singlecast or SVC).

Shared helper functions between PeerConnectionEncodingsIntegrationTest
and PeerConnectionSimulcastTests are placed in a test-only util file.

# Already pass, no need to wait for chromium bots for webrtc testonly CL
NOTRY=True

Bug: webrtc:15063
Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39799}
2023-04-11 07:46:42 +00:00