webrtc_m130

History

Danil Chapovalov 208491c8b9 Revert "Use ArrayView for byte stream parsing in VideoRtpDepacketizerH264"

This reverts commit 4b53e9af6126028497239b39321ec6740f8e2bc2.

Reason for revert: Bug: chromium:371054866

Original change's description:
> Use ArrayView for byte stream parsing in VideoRtpDepacketizerH264
>
>
> Bug: webrtc:42223344, webrtc:42225170
> Change-Id: Ia2025ab225499702c0abe47690742a9c0d6109b7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364380
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43147}

Bug: webrtc:42223344, webrtc:42225170, chromium:371054866
Change-Id: I5c0222add560622a6ce34622d80a4bf7f1fc3fae
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364560
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43155}

2024-10-03 08:52:33 +00:00

async_audio_processing

Deprecate rtc::RefCountInterface

2024-06-07 09:47:26 +00:00

audio_coding

Reland "Delete AcmReceiver"

2024-10-02 13:35:03 +00:00

audio_device

Remove VLA from audio device code.

2024-09-26 15:40:32 +00:00

audio_mixer

Migrate absl::optional to std::optional