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

This is a setup to solve

Browse Source 
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).

BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
minyue@webrtc.org
2014-08-04 14:41:57 +00:00
parent 84b9e1e9d9
commit 0040a6ef97
5 changed files with 110 additions and 16 deletions
Download Patch File Download Diff File Expand all files Collapse all files

28
webrtc/modules/audio_coding/codecs/opus/opus_inst.h Normal file
View File

@ -0,0 +1,28 @@
/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
#include "opus.h"
struct WebRtcOpusEncInst {
OpusEncoder* encoder;
};
struct WebRtcOpusDecInst {
OpusDecoder* decoder_left;
OpusDecoder* decoder_right;
int prev_decoded_samples;
int channels;
};
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_