/external/aac/libAACdec/src/ |
H A D | ldfiltbank.h | 105 const int frameLength
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/external/aac/libAACenc/src/ |
H A D | bandwidth.h | 102 INT frameLength,
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H A D | transform.h | 108 * \param frameLength length of the block. 118 const INT frameLength,
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H A D | transform.cpp | 107 const INT frameLength, 134 tl = frameLength; 140 int offset = (windowShape == LOL_WINDOW) ? ((frameLength * 3)>>2) : 0; 141 fl = frameLength - offset; 142 fr = frameLength - offset; 146 fl = frameLength >> 3; 147 fr = frameLength; 150 fl = frameLength; 151 fr = frameLength >> 3; 154 fl = fr = frameLength >> 102 FDKaacEnc_Transform_Real(const INT_PCM * pTimeData, FIXP_DBL *RESTRICT mdctData, const INT blockType, const INT windowShape, INT *prevWindowShape, const INT frameLength, INT *mdctData_e, INT filterType ,FIXP_DBL * RESTRICT overlapAddBuffer ) argument [all...] |
H A D | bandwidth.cpp | 197 const INT frameLength, 206 switch (frameLength) { 254 switch (frameLength) { 290 INT frameLength, 361 frameLength, 196 GetBandwidthEntry( const INT frameLength, const INT sampleRate, const INT chanBitRate, const INT entryNo) argument 285 FDKaacEnc_DetermineBandWidth(INT* bandWidth, INT proposedBandWidth, INT bitrate, AACENC_BITRATE_MODE bitrateMode, INT sampleRate, INT frameLength, CHANNEL_MAPPING* cm, CHANNEL_MODE encoderMode) argument
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H A D | metadata_main.h | 140 * \param frameLength Number of samples to be processes within one frame. 155 const UINT frameLength,
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H A D | aacenc.h | 229 * \param frameLength the frameLength to be used for the AAC encoder 242 INT frameLength,
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H A D | metadata_main.cpp | 406 const UINT frameLength, 422 for (nFrames=0, delay=audioDelay-frameLength; delay>0; delay-=frameLength, nFrames++); 469 frameLength, 401 FDK_MetadataEnc_Init( HANDLE_FDK_METADATA_ENCODER hMetaData, const INT resetStates, const INT metadataMode, const INT audioDelay, const UINT frameLength, const UINT sampleRate, const UINT nChannels, const CHANNEL_MODE channelMode, const CHANNEL_ORDER channelOrder ) argument
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H A D | aacenc.cpp | 119 INT frameLength, 131 while ( (frameLength & ~((1<<(shift+1))-1)) == frameLength 139 averageBitsPerFrame = (bitRate*(frameLength>>shift)) / (coreSamplingRate>>shift) / nSubFrames; 152 bitRate = FDKmax(bitRate, ((((40 * nChannels) + transportBits) * (coreSamplingRate)) / frameLength) ); 155 bitRate = FDKmin(bitRate, ((nChannelsEff * MIN_BUFSIZE_PER_EFF_CHAN)*(coreSamplingRate>>shift)) / (frameLength>>shift)) ; 116 FDKaacEnc_LimitBitrate( HANDLE_TRANSPORTENC hTpEnc, INT coreSamplingRate, INT frameLength, INT nChannels, INT nChannelsEff, INT bitRate, INT averageBits, INT *pAverageBitsPerFrame, INT bitrateMode, INT nSubFrames ) argument
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H A D | aacenc_lib.cpp | 650 const INT frameLength, 687 frameLength, 1030 INT frameLength = hAacConfig->framelength; local 1040 frameLength = hAacConfig->framelength; /* adapt temporal framelength */ 1100 &frameLength, 1189 frameLength, 1208 hAacEncoder->nSamplesToRead = frameLength * config->nChannels; 2114 pInfo->frameLength = hAacEncoder->nSamplesToRead/hAacEncoder->extParam.nChannels; 647 aacEncoder_LimitBitrate( const HANDLE_TRANSPORTENC hTpEnc, const INT samplingRate, const INT frameLength, const INT nChannels, const CHANNEL_MODE channelMode, INT bitRate, const INT nSubFrames, const INT sbrActive, const INT sbrDownSampleRate, const AUDIO_OBJECT_TYPE aot ) argument
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/external/okhttp/okhttp-ws/src/main/java/com/squareup/okhttp/internal/ws/ |
H A D | WebSocketReader.java | 70 private long frameLength; field in class:WebSocketReader 136 frameLength = b1 & B1_MASK_LENGTH; 137 if (frameLength == PAYLOAD_SHORT) { 138 frameLength = source.readShort() & 0xffffL; // Value is unsigned. 139 } else if (frameLength == PAYLOAD_LONG) { 140 frameLength = source.readLong(); 141 if (frameLength < 0) { 143 "Frame length 0x" + Long.toHexString(frameLength) + " > 0x7FFFFFFFFFFFFFFF"); 148 if (isControlFrame && frameLength > PAYLOAD_MAX) { 160 if (frameBytesRead < frameLength) { [all...] |
/external/apache-harmony/jdwp/src/test/java/org/apache/harmony/jpda/tests/framework/jdwp/ |
H A D | TypesLengths.java | 116 private static int frameLength; field in class:TypesLengths 183 return frameLength; 278 frameLength = typeLength;
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/external/sonivox/arm-fm-22k/lib_src/ |
H A D | eas_data.h | 72 EAS_U32 frameLength; member in struct:s_eas_stream_tag
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H A D | eas_public.c | 323 pStream->frameLength = AUDIO_FRAME_LENGTH; 928 if ((result = EAS_ParseEvents(pEASData, &pEASData->streams[streamNum], pEASData->streams[streamNum].time + pEASData->streams[streamNum].frameLength, eParserModePlay)) != EAS_SUCCESS) 1174 pStream->frameLength = (AUDIO_FRAME_LENGTH * (rate >> 8)) >> 20; 1246 pStream->time += pStream->frameLength;
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/external/sonivox/arm-hybrid-22k/lib_src/ |
H A D | eas_data.h | 72 EAS_U32 frameLength; member in struct:s_eas_stream_tag
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H A D | eas_public.c | 323 pStream->frameLength = AUDIO_FRAME_LENGTH; 928 if ((result = EAS_ParseEvents(pEASData, &pEASData->streams[streamNum], pEASData->streams[streamNum].time + pEASData->streams[streamNum].frameLength, eParserModePlay)) != EAS_SUCCESS) 1174 pStream->frameLength = (AUDIO_FRAME_LENGTH * (rate >> 8)) >> 20; 1246 pStream->time += pStream->frameLength;
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/external/sonivox/arm-wt-22k/lib_src/ |
H A D | eas_data.h | 74 EAS_U32 frameLength; member in struct:s_eas_stream_tag
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H A D | eas_public.c | 323 pStream->frameLength = AUDIO_FRAME_LENGTH; 940 if ((result = EAS_ParseEvents(pEASData, &pEASData->streams[streamNum], pEASData->streams[streamNum].time + pEASData->streams[streamNum].frameLength, eParserModePlay)) != EAS_SUCCESS) 1186 pStream->frameLength = (AUDIO_FRAME_LENGTH * (rate >> 8)) >> 20; 1258 pStream->time += pStream->frameLength;
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/external/aac/libAACenc/include/ |
H A D | aacenc_lib.h | 774 UINT frameLength; /*!< Amount of input audio samples consumed each frame per channel, depending member in struct:__anon57
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/interface/ |
H A D | isacfix.h | 356 * - frameLength : Length of frame in packet (in samples) 361 WebRtc_Word16* frameLength);
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/interface/ |
H A D | isac.h | 319 * - frameLength : Length of frame in packet (in samples) 326 WebRtc_Word16* frameLength); 639 * - frameLength : Length of frame in packet (in samples)
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/external/aac/libSBRenc/include/ |
H A D | sbr_encoder.h | 316 * \param frameLength Input: Encoder frameLength. output core encoder frameLength. 335 INT *frameLength,
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | isac.c | 1692 * - frameLength : Length of frame in packet (in samples) 1740 * - frameLength : Length of frame in packet (in samples) 1745 WebRtc_Word16* frameLength) { 1765 err = WebRtcIsac_DecodeFrameLen(&streamdata, frameLength); 1775 *frameLength <<= 1; 1743 WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst, const WebRtc_Word16* encoded, WebRtc_Word16* frameLength) argument
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/external/aac/libSBRenc/src/ |
H A D | sbr_encoder.cpp | 1817 INT *frameLength, 1831 int coreFrameLength = *frameLength; 2190 *frameLength = coreFrameLength * *downSampleFactor; 2202 *frameLength = coreFrameLength; 1807 sbrEncoder_Init( HANDLE_SBR_ENCODER hSbrEncoder, SBR_ELEMENT_INFO elInfo[(8)], int noElements, INT_PCM *inputBuffer, INT *coreBandwidth, INT *inputBufferOffset, INT *numChannels, INT *coreSampleRate, UINT *downSampleFactor, INT *frameLength, AUDIO_OBJECT_TYPE aot, int *delay, int transformFactor, const int headerPeriod, ULONG statesInitFlag ) argument
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
H A D | isacfix.c | 1257 * - frameLength : Length of frame in packet (in samples) 1262 WebRtc_Word16* frameLength) 1288 err = WebRtcIsacfix_DecodeFrameLen(&streamdata, frameLength); 1305 * - frameLength : Length of frame in packet (in samples) 1261 WebRtcIsacfix_ReadFrameLen(const WebRtc_Word16* encoded, WebRtc_Word16* frameLength) argument
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