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9bb8f0553d
This CL is a followup to https://webrtc-review.googlesource.com/c/src/+/38481. With the new approach we can just use the generic RTP packetizer to pass frames over the wire as the specific info is contained within the bitstream. This makes the new codec more modular and reduces its footprint. Bug: webrtc:7671 Change-Id: Ib07f72a9d338e3cbfdbf39cba9891a959b5f7552 Reviewed-on: https://webrtc-review.googlesource.com/43220 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21753} |
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| .. | ||
| audio_coding | ||
| audio_device | ||
| audio_mixer | ||
| audio_processing | ||
| bitrate_controller | ||
| congestion_controller | ||
| desktop_capture | ||
| include | ||
| media_file | ||
| pacing | ||
| remote_bitrate_estimator | ||
| rtp_rtcp | ||
| utility | ||
| video_capture | ||
| video_coding | ||
| video_processing | ||
| BUILD.gn | ||
| module_common_types_unittest.cc | ||
| OWNERS | ||