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83ed89a45f
This is a backwards-compatible change. It makes WebRTC use the Opus multistream decoder for all Opus packets. Single-stream packets are a special case of multistream ones (with stream=1). The tricky parts are 'WebRtcOpus_GetMaxPlaybackRate' and 'WebRtcOpus_GetSurroundParameters'. GetMaxPlaybackRate is supposed to do what opus_encoder_ctl(encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth)) did when we had single-stream encoders. Now there may be several independent encoders with possibly different BANDWIDTH. The new GetMaxPlaybackRate queries all of them, and returns a playback rate if all the encoder's rates are equal. WebRtcOpus_GetSurroundParameters is a configuration convention. It maps the number of channels to a multi-stream encoder/decoder configuration. As described in RFC 7845 https://tools.ietf.org/html/rfc7845#section-5.1.1, a multi-stream encoder/decoder needs a number of streams, number of coupled streams and a 255-byte mapping array. The function GetSurroundParameters computes all of these from the number of channels. [1, 2, 4, 6, 8] channels are supported. Bug: webrtc:8649 Change-Id: I271de8e387d738254d6aa53af7fcf8644a53edb5 Reviewed-on: https://webrtc-review.googlesource.com/c/111750 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26293} |
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| .. | ||
| cng | ||
| g711 | ||
| g722 | ||
| ilbc | ||
| isac | ||
| opus | ||
| pcm16b | ||
| red | ||
| tools | ||
| audio_decoder.h | ||
| audio_encoder.h | ||
| builtin_audio_decoder_factory_unittest.cc | ||
| builtin_audio_encoder_factory_unittest.cc | ||
| legacy_encoded_audio_frame.cc | ||
| legacy_encoded_audio_frame.h | ||
| legacy_encoded_audio_frame_unittest.cc | ||
| OWNERS | ||