/external/webrtc/webrtc/common_audio/vad/include/ |
H A D | webrtc_vad.h | 65 // - frame_length [i] : Length of audio frame buffer in number of samples. 71 size_t frame_length); 73 // Checks for valid combinations of |rate| and |frame_length|. We support 10, 77 // - frame_length [i] : Speech frame buffer length in number of samples. 80 int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length);
|
/external/webrtc/webrtc/tools/frame_editing/ |
H A D | frame_editing_lib.cc | 40 size_t frame_length = CalcBufferSize(kI420, width, height); local 42 rtc::scoped_ptr<uint8_t[]> temp_buffer(new uint8_t[frame_length]); 56 while ((num_bytes_read = fread(temp_buffer.get(), 1, frame_length, in_fid)) 57 == frame_length) { 61 fwrite(temp_buffer.get(), 1, frame_length, out_fid); 70 fwrite(temp_buffer.get(), 1, frame_length, out_fid); 75 fwrite(temp_buffer.get(), 1, frame_length, out_fid); 80 if (num_bytes_read > 0 && num_bytes_read < frame_length) {
|
/external/webrtc/webrtc/common_audio/vad/ |
H A D | vad_unittest.cc | 30 bool VadTest::ValidRatesAndFrameLengths(int rate, size_t frame_length) { argument 32 if (frame_length == 80 || frame_length == 160 || frame_length == 240) { 37 if (frame_length == 160 || frame_length == 320 || frame_length == 480) { 42 if (frame_length == 320 || frame_length == 640 || frame_length [all...] |
H A D | webrtc_vad.c | 59 size_t frame_length) { 73 if (WebRtcVad_ValidRateAndFrameLength(fs, frame_length) != 0) { 78 vad = WebRtcVad_CalcVad48khz(self, audio_frame, frame_length); 80 vad = WebRtcVad_CalcVad32khz(self, audio_frame, frame_length); 82 vad = WebRtcVad_CalcVad16khz(self, audio_frame, frame_length); 84 vad = WebRtcVad_CalcVad8khz(self, audio_frame, frame_length); 93 int WebRtcVad_ValidRateAndFrameLength(int rate, size_t frame_length) { argument 106 if (frame_length == valid_length) { 58 WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame, size_t frame_length) argument
|
H A D | vad_core.h | 97 * - frame_length : Number of input samples 107 size_t frame_length); 109 size_t frame_length); 111 size_t frame_length); 113 size_t frame_length);
|
H A D | vad_unittest.h | 45 bool ValidRatesAndFrameLengths(int rate, size_t frame_length);
|
/external/libopus/silk/ |
H A D | decoder_set_fs.c | 41 opus_int frame_length, ret = 0; local 48 frame_length = silk_SMULBB( psDec->nb_subfr, psDec->subfr_length ); 58 if( psDec->fs_kHz != fs_kHz || frame_length != psDec->frame_length ) { 100 psDec->frame_length = frame_length; 104 silk_assert( psDec->frame_length > 0 && psDec->frame_length <= MAX_FRAME_LENGTH );
|
H A D | decode_frame.c | 53 L = psDec->frame_length; 75 psDec->indices.quantOffsetType, psDec->frame_length ); 106 silk_assert( psDec->ltp_mem_length >= psDec->frame_length ); 107 mv_len = psDec->ltp_mem_length - psDec->frame_length; 108 silk_memmove( psDec->outBuf, &psDec->outBuf[ psDec->frame_length ], mv_len * sizeof(opus_int16) ); 109 silk_memcpy( &psDec->outBuf[ mv_len ], pOut, psDec->frame_length * sizeof( opus_int16 ) );
|
H A D | decode_pulses.c | 42 const opus_int frame_length /* I Frame length */ 57 iter = silk_RSHIFT( frame_length, LOG2_SHELL_CODEC_FRAME_LENGTH ); 58 if( iter * SHELL_CODEC_FRAME_LENGTH < frame_length ) { 59 silk_assert( frame_length == 12 * 10 ); /* Make sure only happens for 10 ms @ 12 kHz */ 114 silk_decode_signs( psRangeDec, pulses, frame_length, signalType, quantOffsetType, sum_pulses );
|
H A D | stereo_LR_to_MS.c | 47 opus_int frame_length /* I Number of samples */ 61 ALLOC( side, frame_length + 2, opus_int16 ); 63 for( n = 0; n < frame_length + 2; n++ ) { 73 silk_memcpy( state->sMid, &mid[ frame_length ], 2 * sizeof( opus_int16 ) ); 74 silk_memcpy( state->sSide, &side[ frame_length ], 2 * sizeof( opus_int16 ) ); 77 ALLOC( LP_mid, frame_length, opus_int16 ); 78 ALLOC( HP_mid, frame_length, opus_int16 ); 79 for( n = 0; n < frame_length; n++ ) { 86 ALLOC( LP_side, frame_length, opus_int16 ); 87 ALLOC( HP_side, frame_length, opus_int1 [all...] |
H A D | stereo_MS_to_LR.c | 41 opus_int frame_length /* I Number of samples */ 50 silk_memcpy( state->sMid, &x1[ frame_length ], 2 * sizeof( opus_int16 ) ); 51 silk_memcpy( state->sSide, &x2[ frame_length ], 2 * sizeof( opus_int16 ) ); 69 for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) { 79 for( n = 0; n < frame_length; n++ ) {
|
H A D | encode_pulses.c | 65 const opus_int frame_length /* I Frame length */ 87 iter = silk_RSHIFT( frame_length, LOG2_SHELL_CODEC_FRAME_LENGTH ); 88 if( iter * SHELL_CODEC_FRAME_LENGTH < frame_length ) { 89 silk_assert( frame_length == 12 * 10 ); /* Make sure only happens for 10 ms @ 12 kHz */ 91 silk_memset( &pulses[ frame_length ], 0, SHELL_CODEC_FRAME_LENGTH * sizeof(opus_int8)); 204 silk_encode_signs( psRangeEnc, pulses, frame_length, signalType, quantOffsetType, sum_pulses );
|
H A D | VAD.c | 104 silk_assert( MAX_FRAME_LENGTH >= psEncC->frame_length ); 105 silk_assert( psEncC->frame_length <= 512 ); 106 silk_assert( psEncC->frame_length == 8 * silk_RSHIFT( psEncC->frame_length, 3 ) ); 111 decimated_framelength1 = silk_RSHIFT( psEncC->frame_length, 1 ); 112 decimated_framelength2 = silk_RSHIFT( psEncC->frame_length, 2 ); 113 decimated_framelength = silk_RSHIFT( psEncC->frame_length, 3 ); 121 They're arranged to allow the minimal ( frame_length / 4 ) extra 131 X, &X[ X_offset[ 3 ] ], psEncC->frame_length ); 158 decimated_framelength = silk_RSHIFT( psEncC->frame_length, silk_min_in [all...] |
/external/libldac/src/ |
H A D | ldaclib_api.c | 73 int frame_length) 75 if ((0 < frame_length) && (frame_length <= LDAC_MAXNBYTES)) { 84 int frame_length, 88 if ((LDAC_MINSUPNBYTES/2 <= frame_length) && (frame_length <= LDAC_MAXSUPNBYTES/2)) { 96 if ((LDAC_MINSUPNBYTES <= frame_length) && (frame_length <= LDAC_MAXSUPNBYTES)) { 332 int frame_length, 352 if (!ldaclib_assert_frame_length(frame_length)) { 72 ldaclib_assert_frame_length( int frame_length) argument 83 ldaclib_assert_supported_frame_length( int frame_length, int chconfig_id) argument 328 ldaclib_set_config_info( HANDLE_LDAC hData, int smplrate_id, int chconfig_id, int frame_length, int frame_status) argument 401 ldaclib_set_frame_header( HANDLE_LDAC hData, unsigned char *p_stream, int smplrate_id, int chconfig_id, int frame_length, int frame_status) argument 523 ldaclib_set_encode_frame_length( HANDLE_LDAC hData, int frame_length) argument 678 int frame_length; local [all...] |
/external/libvncserver/webclients/novnc/include/ |
H A D | playback.js | 11 var rfb, mode, test_state, frame_idx, frame_length, 24 frame_length = VNC_frame_data.length; 51 while ((frame_idx < frame_length) && (frame.charAt(0) === "}")) { 62 if (frame_idx >= frame_length) {
|
/external/aac/libMpegTPEnc/src/ |
H A D | tpenc_adts.h | 114 USHORT frame_length; member in struct:__anon356 163 int frame_length
|
/external/aac/libAACdec/src/ |
H A D | channel.h | 142 const UINT frame_length,
|
H A D | pulsedata.h | 126 const SHORT frame_length
|
H A D | pulsedata.cpp | 102 const SHORT frame_length 131 if (k >= frame_length) {
|
/external/webrtc/webrtc/tools/agc/ |
H A D | test_utils.cc | 30 const size_t frame_length = local 34 float gain_step = (gain - last_gain) / (frame_length - 1); 35 for (size_t i = 0; i < frame_length; ++i) {
|
/external/libopus/silk/x86/ |
H A D | VAD_sse.c | 68 silk_assert( MAX_FRAME_LENGTH >= psEncC->frame_length ); 69 silk_assert( psEncC->frame_length <= 512 ); 70 silk_assert( psEncC->frame_length == 8 * silk_RSHIFT( psEncC->frame_length, 3 ) ); 75 decimated_framelength1 = silk_RSHIFT( psEncC->frame_length, 1 ); 76 decimated_framelength2 = silk_RSHIFT( psEncC->frame_length, 2 ); 77 decimated_framelength = silk_RSHIFT( psEncC->frame_length, 3 ); 85 They're arranged to allow the minimal ( frame_length / 4 ) extra 95 X, &X[ X_offset[ 3 ] ], psEncC->frame_length ); 122 decimated_framelength = silk_RSHIFT( psEncC->frame_length, silk_min_in [all...] |
/external/webrtc/webrtc/modules/audio_coding/test/ |
H A D | opus_test.h | 37 size_t frame_length,
|
/external/webrtc/webrtc/test/testsupport/metrics/ |
H A D | video_metrics.cc | 111 const size_t frame_length = 3 * width * height >> 1; local 114 rtc::scoped_ptr<uint8_t[]> ref_buffer(new uint8_t[frame_length]); 115 rtc::scoped_ptr<uint8_t[]> test_buffer(new uint8_t[frame_length]); 122 size_t ref_bytes = fread(ref_buffer.get(), 1, frame_length, ref_fp); 123 size_t test_bytes = fread(test_buffer.get(), 1, frame_length, test_fp); 124 while (ref_bytes == frame_length && test_bytes == frame_length) { 147 ref_bytes = fread(ref_buffer.get(), 1, frame_length, ref_fp); 148 test_bytes = fread(test_buffer.get(), 1, frame_length, test_fp);
|
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | bandwidth_estimator.c | 142 const int32_t frame_length, 166 if ( frame_length != bwest_str->prev_frame_length ) 169 1000.0f / (float)frame_length; /* bits/s */ 174 rec_rtp_rate = ((float)pksize * 8.0f * 1000.0f / (float)frame_length) + 187 bwest_str->prev_frame_length = frame_length; 211 if (send_ts_diff <= (16 * frame_length)*2) 225 (float)frame_length); 274 if ( frame_length != bwest_str->prev_frame_length ) 278 1000.0f / (float)frame_length; /* bits/s */ 293 late_diff = arr_ts_diff - (float)(16 * frame_length); 139 WebRtcIsac_UpdateBandwidthEstimator( BwEstimatorstr* bwest_str, const uint16_t rtp_number, const int32_t frame_length, const uint32_t send_ts, const uint32_t arr_ts, const size_t pksize ) argument [all...] |
/external/aac/libMpegTPDec/src/ |
H A D | tpdec_adts.cpp | 203 bs.frame_length = FDKreadBits(hBs, Adts_Length_FrameLength); 223 pAdts->rawDataBlockDist[bs.num_raw_blocks] = bs.frame_length - 7 - bs.num_raw_blocks*2 - 2 ; 255 FDKpushFor(hBs, bs.frame_length * 8); // try again one frame later 265 FDKpushFor(hBs, bs.frame_length * 8); // try again one frame later 271 cmp_buffer_fullness = bs.frame_length*8 + bs.adts_fullness*32*getNumberOfEffectiveChannels(bs.channel_config); 344 FDKpushFor(hBs, (bs.frame_length<<3) - adtsHeaderLength - 3); 369 length = (pAdts->bs.frame_length - 7) << 3; /* aac_frame_length subtracted by the header size (7 bytes). */
|