session-android/jni/webrtc/modules/audio_coding/main/acm2/acm_opus.h
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

67 lines
1.8 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_ACM2_ACM_OPUS_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
struct WebRtcOpusEncInst;
struct WebRtcOpusDecInst;
namespace webrtc {
namespace acm2 {
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) OVERRIDE
EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
virtual int SetFEC(bool enable_fec) OVERRIDE;
virtual int SetPacketLossRate(int loss_rate) OVERRIDE;
virtual int SetOpusMaxBandwidth(int max_bandwidth) OVERRIDE;
protected:
void DestructEncoderSafe();
int16_t InternalCreateEncoder();
void InternalDestructEncoderInst(void* ptr_inst);
int16_t SetBitRateSafe(const int32_t rate) OVERRIDE
EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
WebRtcOpusEncInst* encoder_inst_ptr_;
uint16_t sample_freq_;
int32_t bitrate_;
int channels_;
bool fec_enabled_;
int packet_loss_rate_;
};
} // namespace acm2
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_OPUS_H_