session-android/jni/webrtc/modules/audio_coding/main/acm2/acm_g722.h
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

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

67 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_ACM2_ACM_G722_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_
#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
#include "webrtc/system_wrappers/interface/thread_annotations.h"
typedef struct WebRtcG722EncInst G722EncInst;
typedef struct WebRtcG722DecInst G722DecInst;
namespace webrtc {
namespace acm2 {
// Forward declaration.
struct ACMG722EncStr;
struct ACMG722DecStr;
class ACMG722 : public ACMGenericCodec {
public:
explicit ACMG722(int16_t codec_id);
~ACMG722();
// For FEC.
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);
protected:
int32_t Add10MsDataSafe(const uint32_t timestamp,
const int16_t* data,
const uint16_t length_smpl,
const uint8_t audio_channel)
EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
void DestructEncoderSafe() OVERRIDE
EXCLUSIVE_LOCKS_REQUIRED(codec_wrapper_lock_);
int16_t InternalCreateEncoder();
void InternalDestructEncoderInst(void* ptr_inst);
ACMG722EncStr* ptr_enc_str_;
G722EncInst* encoder_inst_ptr_;
G722EncInst* encoder_inst_ptr_right_; // Prepared for stereo
};
} // namespace acm2
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_