diff options
| author | Amit Hilbuch <amithi@webrtc.org> | 2019-01-17 22:38:48 +0000 |
|---|---|---|
| committer | Commit Bot <commit-bot@chromium.org> | 2019-01-17 22:38:57 +0000 |
| commit | 1fa51d690575c33fa05b33ea78d241e43f7a587a (patch) | |
| tree | fc134b362a9251921385b73ea80cd39736aba149 /resources | |
| parent | 08a9b618a630789759bade26c9308999b5e131d5 (diff) | |
| download | webrtc-1fa51d690575c33fa05b33ea78d241e43f7a587a.tar.gz | |
Revert "Opus multistream."
This reverts commit 83ed89a45f4578ca07efef48e772b9aafb263163.
Reason for revert: breaks downstream project
Original change's description:
> Opus multistream.
>
> 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}
TBR=aleloi@webrtc.org,minyue@webrtc.org
Change-Id: I1002e3273b44d3cccacdba84b8c363eefd537c4b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8649
Reviewed-on: https://webrtc-review.googlesource.com/c/118201
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26306}
Diffstat (limited to 'resources')
| -rw-r--r-- | resources/audio_coding/speech_4_channels_48k_one_second.wav.sha1 | 1 |
1 files changed, 0 insertions, 1 deletions
diff --git a/resources/audio_coding/speech_4_channels_48k_one_second.wav.sha1 b/resources/audio_coding/speech_4_channels_48k_one_second.wav.sha1 deleted file mode 100644 index 7d3041c601..0000000000 --- a/resources/audio_coding/speech_4_channels_48k_one_second.wav.sha1 +++ /dev/null @@ -1 +0,0 @@ -a60c7d03ac2ad9af3cfc7640a4979881f6d47c9c
\ No newline at end of file |