zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Opus multistream.

Browse Source 
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}
This commit is contained in:
Alex Loiko
2019-01-17 12:32:11 +01:00
committed by Commit Bot
parent fe626f53f0
commit 83ed89a45f
7 changed files with 199 additions and 54 deletions
Download Patch File Download Diff File Expand all files Collapse all files

16
modules/audio_coding/codecs/opus/opus_interface.h
View File

@ -125,6 +125,22 @@ 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. */