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> Adding DTX to WebRTC Opus wrapper > > This is a step toward adding Opus DTX support in WebRTC. > > Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See > > https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html > > We transmit the first 1-byte packet to let decoder be in-sync > > BUG=webrtc:1014 > R=henrik.lundin@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/13219004 TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
28 lines
775 B
C
28 lines
775 B
C
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
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#include "opus.h"
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struct WebRtcOpusEncInst {
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OpusEncoder* encoder;
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};
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struct WebRtcOpusDecInst {
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OpusDecoder* decoder;
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int prev_decoded_samples;
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int channels;
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};
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#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
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