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/*
* 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_ISAC_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_
#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/thread_annotations.h"
namespace webrtc {
class CriticalSectionWrapper;
namespace acm2 {
struct ACMISACInst;
enum IsacCodingMode {
ADAPTIVE,
CHANNEL_INDEPENDENT
};
class ACMISAC : public ACMGenericCodec, AudioDecoder {
public:
explicit ACMISAC(int16_t codec_id);
~ACMISAC();
int16_t InternalInitDecoder(WebRtcACMCodecParams* codec_params);
// Methods below are inherited from ACMGenericCodec.
ACMGenericCodec* CreateInstance(void) OVERRIDE;
int16_t InternalEncode(uint8_t* bitstream,
int16_t* bitstream_len_byte) OVERRIDE;
int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params) OVERRIDE;
int16_t UpdateDecoderSampFreq(int16_t codec_id) OVERRIDE;
int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz) OVERRIDE;
int16_t EncoderSampFreq(uint16_t* samp_freq_hz) OVERRIDE;
int32_t ConfigISACBandwidthEstimator(const uint8_t init_frame_size_msec,
const uint16_t init_rate_bit_per_sec,
const bool enforce_frame_size) OVERRIDE;
int32_t SetISACMaxPayloadSize(const uint16_t max_payload_len_bytes) OVERRIDE;
int32_t SetISACMaxRate(const uint32_t max_rate_bit_per_sec) OVERRIDE;
int16_t REDPayloadISAC(const int32_t isac_rate,
const int16_t isac_bw_estimate,
uint8_t* payload,
int16_t* payload_len_bytes) OVERRIDE;
// Methods below are inherited from AudioDecoder.
virtual int Decode(const uint8_t* encoded,
size_t encoded_len,
int16_t* decoded,
SpeechType* speech_type) OVERRIDE;
virtual bool HasDecodePlc() const OVERRIDE { return true; }
virtual int DecodePlc(int num_frames, int16_t* decoded) OVERRIDE;
virtual int Init() OVERRIDE { return 0; }
virtual int IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) OVERRIDE;
virtual int DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int16_t* decoded,
SpeechType* speech_type) OVERRIDE;
virtual int ErrorCode() OVERRIDE;
protected:
int16_t Transcode(uint8_t* bitstream,
int16_t* bitstream_len_byte,
int16_t q_bwe,
int32_t rate,
bool is_red);
void UpdateFrameLen();
// Methods below are inherited from ACMGenericCodec.
void DestructEncoderSafe() OVERRIDE;
int16_t SetBitRateSafe(const int32_t bit_rate) OVERRIDE;
int32_t GetEstimatedBandwidthSafe() OVERRIDE;
int32_t SetEstimatedBandwidthSafe(int32_t estimated_bandwidth) OVERRIDE;
int32_t GetRedPayloadSafe(uint8_t* red_payload,
int16_t* payload_bytes) OVERRIDE;
int16_t InternalCreateEncoder() OVERRIDE;
void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE;
void CurrentRate(int32_t* rate_bit_per_sec) OVERRIDE;
virtual AudioDecoder* Decoder(int codec_id) OVERRIDE;
// |codec_inst_crit_sect_| protects |codec_inst_ptr_|.
const scoped_ptr<CriticalSectionWrapper> codec_inst_crit_sect_;
ACMISACInst* codec_inst_ptr_ GUARDED_BY(codec_inst_crit_sect_);
bool is_enc_initialized_;
IsacCodingMode isac_coding_mode_;
bool enforce_frame_size_;
int32_t isac_current_bn_;
uint16_t samples_in_10ms_audio_;
bool decoder_initialized_;
};
} // namespace acm2
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_
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