webrtc/modules
Mirko Bonadei e1607ed3a6 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}
2024-05-03 08:02:31 +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 Revert "h264: bail out early when failing to parse SPS/PPS ids" 2024-05-03 08:02:31 +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 Query EncoderInfoSettings through propagated field trials 2024-04-30 11:16:31 +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