c454fab03b
TEST=VoE auto-test, audio_coding_module_test; only 15 ms of teststereo_out_1.pcm is not bit-exact with output file of the head revision Review URL: https://webrtc-codereview.appspot.com/937035 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3287 4adac7df-926f-26a2-2b94-8c16560cd09d
71 lines
1.9 KiB
C++
71 lines
1.9 KiB
C++
/*
|
|
* Copyright (c) 2012 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_MAIN_SOURCE_ACM_OPUS_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
|
|
|
|
#include "webrtc/common_audio/resampler/include/resampler.h"
|
|
#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
|
|
|
|
struct WebRtcOpusEncInst;
|
|
struct WebRtcOpusDecInst;
|
|
|
|
namespace webrtc {
|
|
|
|
class ACMOpus : public ACMGenericCodec {
|
|
public:
|
|
explicit ACMOpus(int16_t codec_id);
|
|
~ACMOpus();
|
|
|
|
ACMGenericCodec* CreateInstance(void);
|
|
|
|
int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
|
|
|
|
int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
|
|
|
|
int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
|
|
|
|
protected:
|
|
int16_t DecodeSafe(uint8_t* bitstream,
|
|
int16_t bitstream_len_byte,
|
|
int16_t* audio,
|
|
int16_t* audio_samples,
|
|
int8_t* speech_type);
|
|
|
|
int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
|
|
const CodecInst& codec_inst);
|
|
|
|
void DestructEncoderSafe();
|
|
|
|
void DestructDecoderSafe();
|
|
|
|
int16_t InternalCreateEncoder();
|
|
|
|
int16_t InternalCreateDecoder();
|
|
|
|
void InternalDestructEncoderInst(void* ptr_inst);
|
|
|
|
int16_t SetBitRateSafe(const int32_t rate);
|
|
|
|
bool IsTrueStereoCodec();
|
|
|
|
void SplitStereoPacket(uint8_t* payload, int32_t* payload_length);
|
|
|
|
WebRtcOpusEncInst* encoder_inst_ptr_;
|
|
WebRtcOpusDecInst* decoder_inst_ptr_;
|
|
uint16_t sample_freq_;
|
|
uint16_t bitrate_;
|
|
int channels_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
|