c726b1fc33
BUG=3469 TBR=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6394 4adac7df-926f-26a2-2b94-8c16560cd09d
103 lines
2.8 KiB
C++
103 lines
2.8 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_ACM2_ACM_ISAC_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_
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#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
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namespace webrtc {
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namespace acm2 {
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struct ACMISACInst;
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class AcmAudioDecoderIsac;
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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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~ACMISAC();
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// for FEC
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ACMGenericCodec* CreateInstance(void);
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int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
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int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params);
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int16_t InternalInitDecoder(WebRtcACMCodecParams* codec_params);
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int16_t UpdateDecoderSampFreq(int16_t codec_id);
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int16_t UpdateEncoderSampFreq(uint16_t samp_freq_hz);
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int16_t EncoderSampFreq(uint16_t* samp_freq_hz);
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int32_t ConfigISACBandwidthEstimator(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);
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int32_t SetISACMaxPayloadSize(const uint16_t max_payload_len_bytes);
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int32_t SetISACMaxRate(const uint32_t max_rate_bit_per_sec);
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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);
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protected:
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void DestructEncoderSafe();
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int16_t SetBitRateSafe(const int32_t bit_rate);
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int32_t GetEstimatedBandwidthSafe();
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int32_t SetEstimatedBandwidthSafe(int32_t estimated_bandwidth);
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int32_t GetRedPayloadSafe(uint8_t* red_payload, int16_t* payload_bytes);
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int16_t InternalCreateEncoder();
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void InternalDestructEncoderInst(void* ptr_inst);
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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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void CurrentRate(int32_t* rate_bit_per_sec);
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void UpdateFrameLen();
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virtual AudioDecoder* Decoder(int codec_id);
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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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AcmAudioDecoderIsac* audio_decoder_;
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bool decoder_initialized_;
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};
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} // namespace acm2
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_
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