2d1a55caed
BUG=163 TBR=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1900004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4447 4adac7df-926f-26a2-2b94-8c16560cd09d
135 lines
4.3 KiB
C++
135 lines
4.3 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
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#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
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namespace webrtc {
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struct ACMISACInst;
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enum IsacCodingMode {
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ADAPTIVE,
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CHANNEL_INDEPENDENT
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};
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class ACMISAC : public ACMGenericCodec {
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public:
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explicit ACMISAC(int16_t codec_id);
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virtual ~ACMISAC();
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// for FEC
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virtual ACMGenericCodec* CreateInstance(void) OVERRIDE;
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virtual int16_t InternalEncode(uint8_t* bitstream,
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int16_t* bitstream_len_byte) OVERRIDE;
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virtual int16_t InternalInitEncoder(
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WebRtcACMCodecParams* codec_params) OVERRIDE;
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virtual int16_t InternalInitDecoder(
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WebRtcACMCodecParams* codec_params) OVERRIDE;
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int16_t DeliverCachedIsacData(uint8_t* bitstream,
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int16_t* bitstream_len_byte,
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uint32_t* timestamp,
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WebRtcACMEncodingType* encoding_type,
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const uint16_t isac_rate,
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const uint8_t isac_bwestimate);
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int16_t DeliverCachedData(uint8_t* /* bitstream */,
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int16_t* /* bitstream_len_byte */,
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uint32_t* /* timestamp */,
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WebRtcACMEncodingType* /* encoding_type */) {
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return -1;
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}
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virtual int16_t UpdateDecoderSampFreq(int16_t codec_id) OVERRIDE;
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virtual int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz) OVERRIDE;
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virtual int16_t EncoderSampFreq(uint16_t& samp_freq_hz) OVERRIDE;
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virtual int32_t ConfigISACBandwidthEstimator(
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const uint8_t init_frame_size_msec,
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const uint16_t init_rate_bit_per_sec,
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const bool enforce_frame_size) OVERRIDE;
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virtual int32_t SetISACMaxPayloadSize(
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const uint16_t max_payload_len_bytes) OVERRIDE;
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virtual int32_t SetISACMaxRate(const uint32_t max_rate_bit_per_sec) OVERRIDE;
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virtual int16_t REDPayloadISAC(const int32_t isac_rate,
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const int16_t isac_bw_estimate,
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uint8_t* payload,
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int16_t* payload_len_bytes) OVERRIDE;
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protected:
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virtual int16_t DecodeSafe(uint8_t* bitstream,
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int16_t bitstream_len_byte,
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int16_t* audio,
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int16_t* audio_samples,
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int8_t* speech_type) OVERRIDE;
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virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
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const CodecInst& codec_inst) OVERRIDE;
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virtual void DestructEncoderSafe() OVERRIDE;
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virtual void DestructDecoderSafe() OVERRIDE;
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virtual int16_t SetBitRateSafe(const int32_t bit_rate) OVERRIDE;
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virtual int32_t GetEstimatedBandwidthSafe() OVERRIDE;
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virtual int32_t SetEstimatedBandwidthSafe(
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int32_t estimated_bandwidth) OVERRIDE;
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virtual int32_t GetRedPayloadSafe(uint8_t* red_payload,
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int16_t* payload_bytes) OVERRIDE;
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virtual int16_t InternalCreateEncoder() OVERRIDE;
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virtual int16_t InternalCreateDecoder() OVERRIDE;
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virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE;
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int16_t Transcode(uint8_t* bitstream,
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int16_t* bitstream_len_byte,
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int16_t q_bwe,
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int32_t rate,
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bool is_red);
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virtual void CurrentRate(int32_t& rate_bit_per_sec) OVERRIDE;
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void UpdateFrameLen();
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virtual bool DecoderParamsSafe(WebRtcACMCodecParams *dec_params,
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const uint8_t payload_type) OVERRIDE;
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virtual void SaveDecoderParamSafe(
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const WebRtcACMCodecParams* codec_params) OVERRIDE;
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ACMISACInst* codec_inst_ptr_;
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bool is_enc_initialized_;
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IsacCodingMode isac_coding_mode_;
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bool enforce_frame_size_;
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int32_t isac_current_bn_;
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uint16_t samples_in_10ms_audio_;
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WebRtcACMCodecParams decoder_params_32khz_;
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};
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} // namespace
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
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