48af652ea5
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2. BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2206004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
99 lines
2.8 KiB
C++
99 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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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 webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_
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