webrtc/modules
Danil Chapovalov 8a5f807313 Reland "h264: bail out early when failing to parse SPS/PPS ids"
This reverts commit e1607ed3a6.

Reason for revert: downstream project adjusted

Original change's description:
> Revert "h264: bail out early when failing to parse SPS/PPS ids"
>
> This reverts commit 4344eb713b.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > h264: bail out early when failing to parse SPS/PPS ids
> >
> > This currently gets caught later in the process by the H264 SPS/PPS
> > tracker but can be rejected explicitly here. The network observable
> > behavior should be similar and request a key frame after a 200ms delay, at least for entities that send such bad bitstreams
> >
> > BUG=webrtc:337076010
> >
> > Change-Id: I239c64efa7db631460ef9e9986d283335303df5f
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349060
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Philipp Hancke <phancke@meta.com>
> > Cr-Commit-Position: refs/heads/main@{#42211}
>
> Bug: webrtc:337076010
> Change-Id: I15b815c69f1d25e41fb222d46359655242589fba
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349661
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42217}

Bug: webrtc:337076010
Change-Id: Ibe5a960b9b5fdf9a35e5dfffb47b78ade36b0cec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349700
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42223}
2024-05-03 11:33:45 +00:00
..
async_audio_processing Cleanup rtc::TaskQueue in AsyncAudioProcessing 2024-02-26 12:22:56 +00:00
audio_coding Start using ArrayView in AudioFrame, update PushResampler 2024-04-30 15:33:08 +00:00
audio_device Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
audio_mixer Start using ArrayView in AudioFrame, update PushResampler 2024-04-30 15:33:08 +00:00
audio_processing Adding the option to experiment with the max_allowed_excess_render_blocks parameter. 2024-05-02 12:20:23 +00:00
congestion_controller Limit pacingfactor by upper link capacity estimate. 2024-05-02 15:13:56 +00:00
desktop_capture Fix 'Screen flickering on ScreenCapturerWinDirectx' 2024-04-25 21:18:27 +00:00
include [Unwrap] Delete webrtc::Unwrapper 2023-01-12 14:44:21 +00:00
pacing PacketRouter directly notify RtpTransportControllerSender when sending 2024-03-28 09:27:43 +00:00
portal Video capture PipeWire: add support for DMABuf buffer type 2024-02-27 18:31:26 +00:00
remote_bitrate_estimator Remove expired WebRTC-Bwe-SubtractAdditionalBackoffTerm 2024-04-10 10:11:04 +00:00
rtp_rtcp Reland "h264: bail out early when failing to parse SPS/PPS ids" 2024-05-03 11:33:45 +00:00
third_party [ssci] Added Shipped field to READMEs 2023-07-12 07:31:06 +00:00
utility Rland "Revert "Reland "Reland "Delete old Android ADM."""" 2023-06-30 13:10:12 +00:00
video_capture Deprecate VideoFrame::timestamp() and set_timestamp 2024-03-13 11:08:37 +00:00
video_coding Cleanup unused field trial WebRTC-Video-BandwidthQualityScalerSettings 2024-05-03 10:02:00 +00:00
BUILD.gn [WebRTC-SendPacketsOnWorkerThread] Delete MaybeWorkerThread 2023-04-18 07:07:02 +00:00
module_common_types_unittest.cc [Unwrap] Delete webrtc::Unwrapper 2023-01-12 14:44:21 +00:00