Searched defs:frame_length (Results 1 - 25 of 31) sorted by relevance

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/external/chromium_org/third_party/opus/src/silk/
H A Ddecoder_set_fs.c41 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 );
/external/libopus/silk/
H A Ddecoder_set_fs.c41 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 );
/external/chromium_org/third_party/webrtc/common_audio/vad/
H A Dvad_unittest.cc28 bool VadTest::ValidRatesAndFrameLengths(int rate, int frame_length) { argument
30 if (frame_length == 80 || frame_length == 160 || frame_length == 240) {
35 if (frame_length == 160 || frame_length == 320 || frame_length == 480) {
40 if (frame_length == 320 || frame_length == 640 || frame_length
[all...]
H A Dwebrtc_vad.c72 int frame_length) {
86 if (WebRtcVad_ValidRateAndFrameLength(fs, frame_length) != 0) {
91 vad = WebRtcVad_CalcVad48khz(self, audio_frame, frame_length);
93 vad = WebRtcVad_CalcVad32khz(self, audio_frame, frame_length);
95 vad = WebRtcVad_CalcVad16khz(self, audio_frame, frame_length);
97 vad = WebRtcVad_CalcVad8khz(self, audio_frame, frame_length);
106 int WebRtcVad_ValidRateAndFrameLength(int rate, int frame_length) { argument
119 if (frame_length == valid_length) {
71 WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame, int frame_length) argument
H A Dvad_core.c121 // - frame_length [i] : Number of input samples
125 int16_t total_power, int frame_length) {
149 if (frame_length == 80) {
154 } else if (frame_length == 160) {
607 int frame_length) {
616 int num_10ms_frames = frame_length / kFrameLen10ms48khz;
626 vad = WebRtcVad_CalcVad8khz(inst, speech_nb, frame_length / 6);
632 int frame_length)
641 frame_length);
642 len = WEBRTC_SPL_RSHIFT_W16(frame_length,
124 GmmProbability(VadInstT* self, int16_t* features, int16_t total_power, int frame_length) argument
606 WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame, int frame_length) argument
631 WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame, int frame_length) argument
653 WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame, int frame_length) argument
669 WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame, int frame_length) argument
[all...]
/external/chromium_org/third_party/webrtc/tools/frame_editing/
H A Dframe_editing_lib.cc39 int frame_length = CalcBufferSize(kI420, width, height); local
41 webrtc::scoped_ptr<uint8_t[]> temp_buffer(new uint8_t[frame_length]);
55 while ((num_bytes_read = fread(temp_buffer.get(), 1, frame_length, in_fid))
56 == frame_length) {
60 fwrite(temp_buffer.get(), 1, frame_length, out_fid);
69 fwrite(temp_buffer.get(), 1, frame_length, out_fid);
74 fwrite(temp_buffer.get(), 1, frame_length, out_fid);
79 if (num_bytes_read > 0 && num_bytes_read < frame_length) {
/external/aac/libMpegTPEnc/src/
H A Dtpenc_adts.cpp190 int frame_length)
197 FDK_ASSERT(((frame_length+hAdts->headerBits)/8)<0x2000); /*13 bit*/
233 FDKwriteBits(hBitStream, (frame_length + hAdts->headerBits)>>3, 13);
187 adtsWrite_EncodeHeader(HANDLE_ADTS hAdts, HANDLE_FDK_BITSTREAM hBitStream, int buffer_fullness, int frame_length) argument
H A Dtpenc_adts.h114 USHORT frame_length; member in struct:__anon183
163 int frame_length
/external/chromium_org/media/formats/mpeg/
H A Dadts_stream_parser.cc40 int frame_length; local
53 !reader.ReadBits(13, &frame_length) ||
69 if (sync != 0xfff || layer != 0 || frame_length < bytes_read ||
85 *frame_size = frame_length;
/external/webrtc/src/common_audio/vad/
H A Dvad_unittest.cc40 bool ValidRatesAndFrameLengths(int16_t rate, int16_t frame_length) { argument
42 if (frame_length == 80 || frame_length == 160 || frame_length == 240) {
47 if (frame_length == 160 || frame_length == 320 || frame_length == 480) {
53 if (frame_length == 320 || frame_length == 640 || frame_length
[all...]
H A Dwebrtc_vad.c137 WebRtc_Word16 frame_length)
160 if ((frame_length != 320) && (frame_length != 640) && (frame_length != 960))
164 vad = WebRtcVad_CalcVad32khz((VadInstT*)vad_inst, speech_frame, frame_length);
168 if ((frame_length != 160) && (frame_length != 320) && (frame_length != 480))
172 vad = WebRtcVad_CalcVad16khz((VadInstT*)vad_inst, speech_frame, frame_length);
176 if ((frame_length !
134 WebRtcVad_Process(VadInst *vad_inst, WebRtc_Word16 fs, WebRtc_Word16 *speech_frame, WebRtc_Word16 frame_length) argument
[all...]
/external/aac/libAACdec/src/
H A Dchannel.cpp234 const UINT frame_length,
347 frame_length
228 CChannelElement_Read(HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo[], CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[], const AUDIO_OBJECT_TYPE aot, const SamplingRateInfo *pSamplingRateInfo, const UINT flags, const UINT frame_length, const UCHAR numberOfChannels, const SCHAR epConfig, HANDLE_TRANSPORTDEC pTpDec ) argument
/external/aac/libMpegTPDec/src/
H A Dtpdec_adts.h124 USHORT frame_length; member in struct:__anon168
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
H A Dopus_test.cc211 int frame_length, int percent_loss) {
268 (channels * frame_length);
278 frame_length, kMaxBytes, bitstream);
283 frame_length, kMaxBytes, bitstream);
327 rtp_timestamp_ += frame_length;
328 read_samples += frame_length * channels;
210 Run(TestPackStereo* channel, int channels, int bitrate, int frame_length, int percent_loss) argument
/external/chromium_org/third_party/webrtc/test/testsupport/metrics/
H A Dvideo_metrics.cc111 const size_t frame_length = 3 * width * height >> 1; local
114 scoped_ptr<uint8_t[]> ref_buffer(new uint8_t[frame_length]);
115 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/src/modules/audio_processing/test/testsupport/metrics/
H A Dvideo_metrics.cc108 const int frame_length = 3 * width * height >> 1; local
109 uint8_t* ref = new uint8_t[frame_length];
110 uint8_t* test = new uint8_t[frame_length];
112 int ref_bytes = fread(ref, 1, frame_length, ref_fp);
113 int test_bytes = fread(test, 1, frame_length, test_fp);
114 while (ref_bytes == frame_length && test_bytes == frame_length) {
134 ref_bytes = fread(ref, 1, frame_length, ref_fp);
135 test_bytes = fread(test, 1, frame_length, test_fp);
/external/webrtc/test/testsupport/metrics/
H A Dvideo_metrics.cc108 const int frame_length = 3 * width * height >> 1; local
109 uint8_t* ref = new uint8_t[frame_length];
110 uint8_t* test = new uint8_t[frame_length];
112 int ref_bytes = fread(ref, 1, frame_length, ref_fp);
113 int test_bytes = fread(test, 1, frame_length, test_fp);
114 while (ref_bytes == frame_length && test_bytes == frame_length) {
134 ref_bytes = fread(ref, 1, frame_length, ref_fp);
135 test_bytes = fread(test, 1, frame_length, test_fp);
/external/chromium_org/content/browser/speech/
H A Daudio_encoder.cc163 int frame_length = speex_bits_write(&bits_, encoded_frame_data_ + 1, local
165 encoded_frame_data_[0] = static_cast<char>(frame_length);
167 reinterpret_cast<uint8*>(&encoded_frame_data_[0]), frame_length + 1); local
/external/chromium_org/net/websockets/
H A Dwebsocket_frame_parser_test.cc33 uint64 frame_length; member in struct:net::__anon9215::FrameHeaderTestCase
109 size_t frame_length; member in struct:net::__anon9215::Input
136 kInputs[i].frame + kInputs[i].frame_length);
311 uint64 frame_length = kFrameHeaderTests[i].frame_length; local
316 uint64 input_payload_size = std::min(frame_length, kMaxPayloadSize);
336 if (frame_length == input_payload_size) {
361 EXPECT_EQ(frame_length, header->payload_length);
369 uint64 frame_length = kFrameHeaderTests[i].frame_length; local
[all...]
H A Dwebsocket_frame_test.cc35 uint64 frame_length; member in struct:net::TestCase
51 header.payload_length = kTests[i].frame_length;
73 uint64 frame_length; member in struct:net::TestCase
96 header.payload_length = kTests[i].frame_length;
/external/chromium_org/third_party/opus/src/silk/float/
H A Dpitch_analysis_core_FLP.c104 opus_int frame_length, frame_length_8kHz, frame_length_4kHz; local
122 frame_length = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * Fs_kHz;
139 silk_float2short_array( frame_16_FIX, frame, frame_length );
141 silk_resampler_down2( filt_state, frame_8_FIX, frame_16_FIX, frame_length );
146 silk_float2short_array( frame_12_FIX, frame, frame_length );
148 silk_resampler_down2_3( filt_state, frame_8_FIX, frame_12_FIX, frame_length );
/external/libopus/silk/float/
H A Dpitch_analysis_core_FLP.c104 opus_int frame_length, frame_length_8kHz, frame_length_4kHz; local
122 frame_length = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * Fs_kHz;
139 silk_float2short_array( frame_16_FIX, frame, frame_length );
141 silk_resampler_down2( filt_state, frame_8_FIX, frame_16_FIX, frame_length );
146 silk_float2short_array( frame_12_FIX, frame, frame_length );
148 silk_resampler_down2_3( filt_state, frame_8_FIX, frame_12_FIX, frame_length );
/external/chromium_org/third_party/opus/src/silk/fixed/
H A Dpitch_analysis_core_FIX.c112 opus_int frame_length, frame_length_8kHz, frame_length_4kHz; local
132 frame_length = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * Fs_kHz;
143 silk_resampler_down2( filt_state, frame_8kHz, frame, frame_length );
146 silk_resampler_down2_3( filt_state, frame_8kHz, frame, frame_length );
469 silk_sum_sqr_shift( &energy, &shift, frame, frame_length );
470 ALLOC( scratch_mem, shift > 0 ? frame_length : ALLOC_NONE, opus_int16 );
474 for( i = 0; i < frame_length; i++ ) {
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/
H A Dbandwidth_estimator.c137 const int32_t frame_length,
160 if ( frame_length != bwest_str->prev_frame_length )
163 1000.0f / (float)frame_length; /* bits/s */
168 rec_rtp_rate = ((float)pksize * 8.0f * 1000.0f / (float)frame_length) +
181 bwest_str->prev_frame_length = frame_length;
205 if (send_ts_diff <= (16 * frame_length)*2)
219 (float)frame_length);
268 if ( frame_length != bwest_str->prev_frame_length )
272 1000.0f / (float)frame_length; /* bits/s */
287 late_diff = arr_ts_diff - (float)(16 * frame_length);
134 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 int32_t pksize ) argument
[all...]
/external/libopus/silk/fixed/
H A Dpitch_analysis_core_FIX.c112 opus_int frame_length, frame_length_8kHz, frame_length_4kHz; local
132 frame_length = ( PE_LTP_MEM_LENGTH_MS + nb_subfr * PE_SUBFR_LENGTH_MS ) * Fs_kHz;
143 silk_resampler_down2( filt_state, frame_8kHz, frame, frame_length );
146 silk_resampler_down2_3( filt_state, frame_8kHz, frame, frame_length );
469 silk_sum_sqr_shift( &energy, &shift, frame, frame_length );
470 ALLOC( scratch_mem, shift > 0 ? frame_length : ALLOC_NONE, opus_int16 );
474 for( i = 0; i < frame_length; i++ ) {

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