c454fab03b
TEST=VoE auto-test, audio_coding_module_test; only 15 ms of teststereo_out_1.pcm is not bit-exact with output file of the head revision Review URL: https://webrtc-codereview.appspot.com/937035 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3287 4adac7df-926f-26a2-2b94-8c16560cd09d
131 lines
4.3 KiB
C++
131 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(WebRtc_Word16 codec_id);
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~ACMISAC();
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// for FEC
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ACMGenericCodec* CreateInstance(void);
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WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
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WebRtc_Word16* bitstream_len_byte);
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WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codec_params);
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WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codec_params);
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WebRtc_Word16 DeliverCachedIsacData(WebRtc_UWord8* bitstream,
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WebRtc_Word16* bitstream_len_byte,
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WebRtc_UWord32* timestamp,
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WebRtcACMEncodingType* encoding_type,
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const WebRtc_UWord16 isac_rate,
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const WebRtc_UWord8 isac_bwestimate);
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WebRtc_Word16 DeliverCachedData(WebRtc_UWord8* /* bitstream */,
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WebRtc_Word16* /* bitstream_len_byte */,
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WebRtc_UWord32* /* timestamp */,
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WebRtcACMEncodingType* /* encoding_type */) {
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return -1;
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}
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WebRtc_Word16 UpdateDecoderSampFreq(WebRtc_Word16 codec_id);
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WebRtc_Word16 UpdateEncoderSampFreq(WebRtc_UWord16 samp_freq_hz);
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WebRtc_Word16 EncoderSampFreq(WebRtc_UWord16& samp_freq_hz);
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WebRtc_Word32 ConfigISACBandwidthEstimator(
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const WebRtc_UWord8 init_frame_size_msec,
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const WebRtc_UWord16 init_rate_bit_per_sec,
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const bool enforce_frame_size);
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WebRtc_Word32 SetISACMaxPayloadSize(
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const WebRtc_UWord16 max_payload_len_bytes);
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WebRtc_Word32 SetISACMaxRate(const WebRtc_UWord32 max_rate_bit_per_sec);
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WebRtc_Word16 REDPayloadISAC(const WebRtc_Word32 isac_rate,
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const WebRtc_Word16 isac_bw_estimate,
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WebRtc_UWord8* payload,
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WebRtc_Word16* payload_len_bytes);
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protected:
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WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitstream,
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WebRtc_Word16 bitstream_len_byte,
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WebRtc_Word16* audio,
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WebRtc_Word16* audio_samples,
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WebRtc_Word8* speech_type);
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WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codec_def,
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const CodecInst& codec_inst);
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void DestructEncoderSafe();
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void DestructDecoderSafe();
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WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 bit_rate);
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WebRtc_Word32 GetEstimatedBandwidthSafe();
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WebRtc_Word32 SetEstimatedBandwidthSafe(WebRtc_Word32 estimated_bandwidth);
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WebRtc_Word32 GetRedPayloadSafe(WebRtc_UWord8* red_payload,
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WebRtc_Word16* payload_bytes);
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WebRtc_Word16 InternalCreateEncoder();
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WebRtc_Word16 InternalCreateDecoder();
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void InternalDestructEncoderInst(void* ptr_inst);
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WebRtc_Word16 Transcode(WebRtc_UWord8* bitstream,
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WebRtc_Word16* bitstream_len_byte,
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WebRtc_Word16 q_bwe,
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WebRtc_Word32 rate,
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bool is_red);
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void CurrentRate(WebRtc_Word32& rate_bit_per_sec);
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void UpdateFrameLen();
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bool DecoderParamsSafe(WebRtcACMCodecParams *dec_params,
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const WebRtc_UWord8 payload_type);
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void SaveDecoderParamSafe(const WebRtcACMCodecParams* codec_params);
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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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WebRtc_Word32 isac_current_bn_;
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WebRtc_UWord16 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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