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Revert "Opus multistream."

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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}
This commit is contained in:
Amit Hilbuch
2019-01-17 22:38:48 +00:00
committed by Commit Bot
parent 08a9b618a6
commit 1fa51d6905
7 changed files with 54 additions and 199 deletions
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16
modules/audio_coding/codecs/opus/opus_interface.h
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@ -125,22 +125,6 @@ int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate);
*/
int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz);
/****************************************************************************
* WebRtcOpus_GetMaxPlaybackRate(...)
*
* Queries the maximum playback rate for encoding. If different single-stream
* encoders have different maximum playback rates, this function fails.
*
* Input:
* - inst : Encoder context.
* Output:
* - result_hz : The maximum playback rate in Hz.
* Return value : 0 - Success
* -1 - Error
*/
int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
int32_t* result_hz);
/* TODO(minyue): Check whether an API to check the FEC and the packet loss rate
* is needed. It might not be very useful since there are not many use cases and
* the caller can always maintain the states. */