blob: 20b6c5391be417a77f2118d5ca09ac2aeb45274d [file] [log] [blame]
/*
* 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_ISAC_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
namespace webrtc {
namespace acm1 {
struct ACMISACInst;
enum IsacCodingMode {
ADAPTIVE,
CHANNEL_INDEPENDENT
};
class ACMISAC : public ACMGenericCodec {
public:
explicit ACMISAC(int16_t codec_id);
virtual ~ACMISAC();
// for FEC
virtual ACMGenericCodec* CreateInstance(void) OVERRIDE;
virtual int16_t InternalEncode(uint8_t* bitstream,
int16_t* bitstream_len_byte) OVERRIDE;
virtual int16_t InternalInitEncoder(
WebRtcACMCodecParams* codec_params) OVERRIDE;
virtual int16_t InternalInitDecoder(
WebRtcACMCodecParams* codec_params) OVERRIDE;
int16_t DeliverCachedIsacData(uint8_t* bitstream,
int16_t* bitstream_len_byte,
uint32_t* timestamp,
WebRtcACMEncodingType* encoding_type,
const uint16_t isac_rate,
const uint8_t isac_bwestimate);
int16_t DeliverCachedData(uint8_t* /* bitstream */,
int16_t* /* bitstream_len_byte */,
uint32_t* /* timestamp */,
WebRtcACMEncodingType* /* encoding_type */) {
return -1;
}
virtual int16_t UpdateDecoderSampFreq(int16_t codec_id) OVERRIDE;
virtual int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz) OVERRIDE;
virtual int16_t EncoderSampFreq(uint16_t& samp_freq_hz) OVERRIDE;
virtual int32_t ConfigISACBandwidthEstimator(
const uint8_t init_frame_size_msec,
const uint16_t init_rate_bit_per_sec,
const bool enforce_frame_size) OVERRIDE;
virtual int32_t SetISACMaxPayloadSize(
const uint16_t max_payload_len_bytes) OVERRIDE;
virtual int32_t SetISACMaxRate(const uint32_t max_rate_bit_per_sec) OVERRIDE;
virtual int16_t REDPayloadISAC(const int32_t isac_rate,
const int16_t isac_bw_estimate,
uint8_t* payload,
int16_t* payload_len_bytes) OVERRIDE;
protected:
virtual int16_t DecodeSafe(uint8_t* bitstream,
int16_t bitstream_len_byte,
int16_t* audio,
int16_t* audio_samples,
int8_t* speech_type) OVERRIDE;
virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
const CodecInst& codec_inst) OVERRIDE;
virtual void DestructEncoderSafe() OVERRIDE;
virtual void DestructDecoderSafe() OVERRIDE;
virtual int16_t SetBitRateSafe(const int32_t bit_rate) OVERRIDE;
virtual int32_t GetEstimatedBandwidthSafe() OVERRIDE;
virtual int32_t SetEstimatedBandwidthSafe(
int32_t estimated_bandwidth) OVERRIDE;
virtual int32_t GetRedPayloadSafe(uint8_t* red_payload,
int16_t* payload_bytes) OVERRIDE;
virtual int16_t InternalCreateEncoder() OVERRIDE;
virtual int16_t InternalCreateDecoder() OVERRIDE;
virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE;
int16_t Transcode(uint8_t* bitstream,
int16_t* bitstream_len_byte,
int16_t q_bwe,
int32_t rate,
bool is_red);
virtual void CurrentRate(int32_t& rate_bit_per_sec) OVERRIDE;
void UpdateFrameLen();
virtual bool DecoderParamsSafe(WebRtcACMCodecParams* dec_params,
const uint8_t payload_type) OVERRIDE;
virtual void SaveDecoderParamSafe(
const WebRtcACMCodecParams* codec_params) OVERRIDE;
ACMISACInst* codec_inst_ptr_;
bool is_enc_initialized_;
IsacCodingMode isac_coding_mode_;
bool enforce_frame_size_;
int32_t isac_current_bn_;
uint16_t samples_in_10ms_audio_;
WebRtcACMCodecParams decoder_params_32khz_;
};
} // namespace acm1
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_