fb389e3b02
Patch Set 1: Removing blanks at end of lines. Patch Set 2: Removing tabs. Patch Set 3: Fixing include-guards. Patch Set 4-7: Formatting files in the list. Patch Set 8: Formatting CNG. Patch Set 9: * Fixing comments from code review * Fixing formating in acm_dtmf_playout.cc * Started fixing formating of acm_g7221.cc. More work needed, so don't spend too much time reviewing. * Refactored constructor of ACMGenericCodec. Rest of file still to be fixed. * Fixing break; after return ...; in several files. Patch Set 10: * Chaning from reintepret_cast to static_cast in three files, acm_amr.cc, acm_cng.cc and acm_g722.cc NOTE! Not all files have the right format. That work will continue in separate CLs. Review URL: http://webrtc-codereview.appspot.com/175002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@881 4adac7df-926f-26a2-2b94-8c16560cd09d
154 lines
4.2 KiB
C++
154 lines
4.2 KiB
C++
/*
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* Copyright (c) 2011 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 "acm_generic_codec.h"
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namespace webrtc
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{
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struct ACMISACInst;
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enum iSACCodingMode {ADAPTIVE, CHANNEL_INDEPENDENT};
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class ACMISAC : public ACMGenericCodec
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{
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public:
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ACMISAC(WebRtc_Word16 codecID);
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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(
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WebRtc_UWord8* bitstream,
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WebRtc_Word16* bitStreamLenByte);
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WebRtc_Word16 InternalInitEncoder(
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WebRtcACMCodecParams *codecParams);
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WebRtc_Word16 InternalInitDecoder(
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WebRtcACMCodecParams *codecParams);
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WebRtc_Word16 DeliverCachedIsacData(
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WebRtc_UWord8* bitStream,
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WebRtc_Word16* bitStreamLenByte,
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WebRtc_UWord32* timestamp,
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WebRtcACMEncodingType* encodingType,
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const WebRtc_UWord16 isacRate,
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const WebRtc_UWord8 isacBWestimate);
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WebRtc_Word16 DeliverCachedData(
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WebRtc_UWord8* /* bitStream */,
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WebRtc_Word16* /* bitStreamLenByte */,
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WebRtc_UWord32* /* timestamp */,
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WebRtcACMEncodingType* /* encodingType */)
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{
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return -1;
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}
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WebRtc_Word16 UpdateDecoderSampFreq(
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WebRtc_Word16 codecId);
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WebRtc_Word16 UpdateEncoderSampFreq(
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WebRtc_UWord16 sampFreqHz);
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WebRtc_Word16 EncoderSampFreq(
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WebRtc_UWord16& sampFreqHz);
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WebRtc_Word32 ConfigISACBandwidthEstimator(
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const WebRtc_UWord8 initFrameSizeMsec,
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const WebRtc_UWord16 initRateBitPerSec,
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const bool enforceFrameSize);
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WebRtc_Word32 SetISACMaxPayloadSize(
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const WebRtc_UWord16 maxPayloadLenBytes);
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WebRtc_Word32 SetISACMaxRate(
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const WebRtc_UWord32 maxRateBitPerSec);
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WebRtc_Word16 REDPayloadISAC(
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const WebRtc_Word32 isacRate,
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const WebRtc_Word16 isacBwEstimate,
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WebRtc_UWord8* payload,
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WebRtc_Word16* payloadLenBytes);
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protected:
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WebRtc_Word16 DecodeSafe(
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WebRtc_UWord8* bitStream,
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WebRtc_Word16 bitStreamLenByte,
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WebRtc_Word16* audio,
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WebRtc_Word16* audioSamples,
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WebRtc_Word8* speechType);
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WebRtc_Word32 CodecDef(
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WebRtcNetEQ_CodecDef& codecDef,
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const CodecInst& codecInst);
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void DestructEncoderSafe();
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void DestructDecoderSafe();
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WebRtc_Word16 SetBitRateSafe(
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const WebRtc_Word32 bitRate);
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WebRtc_Word32 GetEstimatedBandwidthSafe();
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WebRtc_Word32 SetEstimatedBandwidthSafe(WebRtc_Word32 estimatedBandwidth);
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WebRtc_Word32 GetRedPayloadSafe(
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WebRtc_UWord8* redPayload,
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WebRtc_Word16* payloadBytes);
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WebRtc_Word16 InternalCreateEncoder();
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WebRtc_Word16 InternalCreateDecoder();
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void InternalDestructEncoderInst(
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void* ptrInst);
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WebRtc_Word16 Transcode(
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WebRtc_UWord8* bitStream,
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WebRtc_Word16* bitStreamLenByte,
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WebRtc_Word16 qBWE,
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WebRtc_Word32 rate,
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bool isRED);
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WebRtc_Word16 UnregisterFromNetEqSafe(
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ACMNetEQ* netEq,
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WebRtc_Word16 payloadType);
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void CurrentRate(WebRtc_Word32& rateBitPerSec);
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void UpdateFrameLen();
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bool DecoderParamsSafe(
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WebRtcACMCodecParams *decParams,
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const WebRtc_UWord8 payloadType);
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void SaveDecoderParamSafe(
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const WebRtcACMCodecParams* codecParams);
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ACMISACInst* _codecInstPtr;
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bool _isEncInitialized;
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iSACCodingMode _isacCodingMode;
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bool _enforceFrameSize;
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WebRtc_Word32 _isacCurrentBN;
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WebRtc_UWord16 _samplesIn10MsAudio;
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WebRtcACMCodecParams _decoderParams32kHz;
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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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