0040a6ef97
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
29 lines
810 B
C
29 lines
810 B
C
/*
|
|
* 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_
|