/external/chromium_org/third_party/opus/src/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 );
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/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 );
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/external/chromium_org/third_party/webrtc/common_audio/vad/ |
H A D | vad_unittest.cc | 28 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 D | webrtc_vad.c | 72 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
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H A D | vad_core.c | 121 // - 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 D | frame_editing_lib.cc | 39 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) {
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/external/aac/libMpegTPEnc/src/ |
H A D | tpenc_adts.cpp | 190 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
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H A D | tpenc_adts.h | 114 USHORT frame_length; member in struct:__anon183 163 int frame_length
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/external/chromium_org/media/formats/mpeg/ |
H A D | adts_stream_parser.cc | 40 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;
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/external/webrtc/src/common_audio/vad/ |
H A D | vad_unittest.cc | 40 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 D | webrtc_vad.c | 137 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 D | channel.cpp | 234 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
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/external/aac/libMpegTPDec/src/ |
H A D | tpdec_adts.h | 124 USHORT frame_length; member in struct:__anon168
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | opus_test.cc | 211 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
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/external/chromium_org/third_party/webrtc/test/testsupport/metrics/ |
H A D | video_metrics.cc | 111 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);
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/external/webrtc/src/modules/audio_processing/test/testsupport/metrics/ |
H A D | video_metrics.cc | 108 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);
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/external/webrtc/test/testsupport/metrics/ |
H A D | video_metrics.cc | 108 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);
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/external/chromium_org/content/browser/speech/ |
H A D | audio_encoder.cc | 163 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
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/external/chromium_org/net/websockets/ |
H A D | websocket_frame_parser_test.cc | 33 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 D | websocket_frame_test.cc | 35 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;
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/external/chromium_org/third_party/opus/src/silk/float/ |
H A D | pitch_analysis_core_FLP.c | 104 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 );
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/external/libopus/silk/float/ |
H A D | pitch_analysis_core_FLP.c | 104 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 );
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/external/chromium_org/third_party/opus/src/silk/fixed/ |
H A D | pitch_analysis_core_FIX.c | 112 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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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | bandwidth_estimator.c | 137 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 D | pitch_analysis_core_FIX.c | 112 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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