/external/chromium_org/third_party/libjingle/source/talk/base/ |
H A D | bandwidthsmoother.cc | 49 // Samples a new bandwidth measurement 50 // returns true if the bandwidth estimation changed 51 bool BandwidthSmoother::Sample(uint32 sample_time, int bandwidth) { argument 52 if (bandwidth < 0) { 56 accumulator_.AddSample(bandwidth); 64 // Replace bandwidth with the mean of sampled bandwidths. 76 // If bandwidth goes any higher we would overflow. 90 // positive bandwidth means we have regained connectivity.
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H A D | virtualsocketserver.h | 57 // Limits the network bandwidth (maximum bytes per second). Zero means that 59 uint32 bandwidth() const { return bandwidth_; } function in class:talk_base::VirtualSocketServer 60 void set_bandwidth(uint32 bandwidth) { bandwidth_ = bandwidth; } argument
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H A D | virtualsocket_unittest.cc | 90 : thread(th), socket(new AsyncUDPSocket(s)), bandwidth(bw), done(false), 125 if (bandwidth > 0) 126 ASSERT_TRUE(sec_count <= 5 * bandwidth / 4); 133 uint32 bandwidth; member in struct:Receiver 636 uint32 bandwidth = 64 * 1024; local 637 ss_->set_bandwidth(bandwidth); 641 Receiver receiver(pthMain, recv_socket, bandwidth); 647 ASSERT_TRUE(receiver.count >= 5 * 3 * bandwidth / 4); 648 ASSERT_TRUE(receiver.count <= 6 * bandwidth); // queue could drain for 1s
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/external/chromium_org/third_party/webrtc/base/ |
H A D | bandwidthsmoother.cc | 32 // Samples a new bandwidth measurement 33 // returns true if the bandwidth estimation changed 34 bool BandwidthSmoother::Sample(uint32 sample_time, int bandwidth) { argument 35 if (bandwidth < 0) { 39 accumulator_.AddSample(bandwidth); 47 // Replace bandwidth with the mean of sampled bandwidths. 59 // If bandwidth goes any higher we would overflow. 73 // positive bandwidth means we have regained connectivity.
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H A D | virtualsocketserver.h | 40 // Limits the network bandwidth (maximum bytes per second). Zero means that 42 uint32 bandwidth() const { return bandwidth_; } function in class:rtc::VirtualSocketServer 43 void set_bandwidth(uint32 bandwidth) { bandwidth_ = bandwidth; } argument
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H A D | virtualsocket_unittest.cc | 73 : thread(th), socket(new AsyncUDPSocket(s)), bandwidth(bw), done(false), 108 if (bandwidth > 0) 109 ASSERT_TRUE(sec_count <= 5 * bandwidth / 4); 116 uint32 bandwidth; member in struct:Receiver 619 uint32 bandwidth = 64 * 1024; local 620 ss_->set_bandwidth(bandwidth); 624 Receiver receiver(pthMain, recv_socket, bandwidth); 630 ASSERT_TRUE(receiver.count >= 5 * 3 * bandwidth / 4); 631 ASSERT_TRUE(receiver.count <= 6 * bandwidth); // queue could drain for 1s
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/external/chromium_org/net/base/ |
H A D | bandwidth_metrics.h | 16 // Tracks statistics about the bandwidth metrics over time. In order to 20 // bandwidth, but not both. 24 // progress concurrently, you have to look at the aggregate bandwidth at any 29 // We can't measure bandwidth by looking at any individual stream. 30 // We can only measure actual bandwidth by looking at the bandwidth 63 // Get the bandwidth. Returns Kbps (kilo-bits-per-second). 64 double bandwidth() const { function in class:net::BandwidthMetrics 81 // We don't use small streams when tracking bandwidth because they are not 96 << "Kbps (avg " << bandwidth() << "Kbp [all...] |
/external/tcpdump/ |
H A D | print-igrp.c | 47 register u_int delay, bandwidth; local 61 bandwidth = EXTRACT_24BITS(igr->igr_bw); 62 metric = bandwidth + delay; 68 10 * delay, bandwidth == 0 ? 0 : 10000000 / bandwidth,
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/external/aac/libAACenc/src/ |
H A D | bandwidth.cpp | 14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by 87 contents/description: bandwidth expert 92 #include "bandwidth.h" 202 INT bandwidth = -1; local 256 bandwidth = (entryNo==0) 270 bandwidth = (INT)scaleValue(fMult(bwFac_fix, (FIXP_DBL)(endBw-startBw)),q_res) + startBw; 274 bandwidth = -1; 281 return bandwidth; 332 /* bandwidth limiting */ 336 else { /* search reasonable bandwidth */ [all...] |
H A D | psy_configuration.cpp | 14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by 570 INT bandwidth, 587 psyConf->allowIS = (useIS) && ( (bitrate/bandwidth) < 5 ); 623 psyConf->lowpassLine = (INT)((2*bandwidth*frameLengthLong)/samplerate); 627 psyConf->lowpassLine = (INT)((2*bandwidth*frameLengthShort)/samplerate); 568 FDKaacEnc_InitPsyConfiguration(INT bitrate, INT samplerate, INT bandwidth, INT blocktype, INT granuleLength, INT useIS, PSY_CONFIGURATION *psyConf, FB_TYPE filterbank) argument
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H A D | psy_main.cpp | 14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by 306 INT bandwidth, 335 ErrorStatus = FDKaacEnc_InitPsyConfiguration(bitRate/channelsEff, sampleRate, bandwidth, LONG_WINDOW, hPsy->granuleLength, useIS, &(hPsy->psyConf[0]), filterBank); 355 ErrorStatus = FDKaacEnc_InitPsyConfiguration(bitRate/channelsEff, sampleRate, bandwidth, SHORT_WINDOW, hPsy->granuleLength, useIS, &hPsy->psyConf[1], filterBank); 299 FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy, AUDIO_OBJECT_TYPE audioObjectType, CHANNEL_MAPPING *cm, INT sampleRate, INT granuleLength, INT bitRate, INT tnsMask, INT bandwidth, INT usePns, INT useIS, UINT syntaxFlags, ULONG initFlags) argument
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/external/chromium_org/net/quic/congestion_control/ |
H A D | send_algorithm_simulator.cc | 29 QuicBandwidth bandwidth, 42 bandwidth_(bandwidth), 57 // average bandwidth in bytes per second. The time elapsed is based on 25 SendAlgorithmSimulator( SendAlgorithmInterface* send_algorithm, MockClock* clock, RttStats* rtt_stats, QuicBandwidth bandwidth, QuicTime::Delta rtt) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | encode_lpc_swb.c | 39 * -bandwidth : indicates if the given LAR vectors belong 50 int16_t bandwidth) 56 switch(bandwidth) 95 * -bandwidth : indicates if the given LAR vectors belong 105 int16_t bandwidth) 114 switch(bandwidth) 169 * -bandwidth : indicates if the given LAR vectors belong 179 int16_t bandwidth) 187 switch(bandwidth) 237 * -bandwidth 48 WebRtcIsac_RemoveLarMean( double* lar, int16_t bandwidth) argument 102 WebRtcIsac_DecorrelateIntraVec( const double* data, double* out, int16_t bandwidth) argument 176 WebRtcIsac_DecorrelateInterVec( const double* data, double* out, int16_t bandwidth) argument 245 WebRtcIsac_QuantizeUncorrLar( double* data, int* recIdx, int16_t bandwidth) argument 315 WebRtcIsac_DequantizeLpcParam( const int* idx, double* out, int16_t bandwidth) argument 371 WebRtcIsac_CorrelateIntraVec( const double* data, double* out, int16_t bandwidth) argument 434 WebRtcIsac_CorrelateInterVec( const double* data, double* out, int16_t bandwidth) argument 499 WebRtcIsac_AddLarMean( double* data, int16_t bandwidth) argument [all...] |
H A D | lpc_analysis.c | 305 /* bandwidth expansion */ 332 /* bandwidth expansion */ 373 * -bandwidth : specifies if the codec is in 0-16 kHz mode or 391 int16_t bandwidth) 396 int16_t numSubFrames = SUBFRAMES * (1 + (bandwidth == isac16kHz)); 442 (bandwidth == isac12kHz); 444 (bandwidth == isac16kHz); 453 /* bandwidth expansion */ 385 WebRtcIsac_GetLpcCoefUb( double* inSignal, MaskFiltstr* maskdata, double* lpCoeff, double corrMat[][UB_LPC_ORDER + 1], double* varscale, int16_t bandwidth) argument
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/external/chromium_org/third_party/webrtc/modules/bitrate_controller/ |
H A D | send_side_bandwidth_estimation.cc | 89 void SendSideBandwidthEstimation::UpdateReceiverEstimate(uint32_t bandwidth) { argument 90 bwe_incoming_ = bandwidth; 206 LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate_ / 1000
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | encode_lpc_swb.c | 39 * -bandwidth : indicates if the given LAR vectors belong 50 WebRtc_Word16 bandwidth) 56 switch(bandwidth) 95 * -bandwidth : indicates if the given LAR vectors belong 105 WebRtc_Word16 bandwidth) 114 switch(bandwidth) 169 * -bandwidth : indicates if the given LAR vectors belong 179 WebRtc_Word16 bandwidth) 187 switch(bandwidth) 237 * -bandwidth 48 WebRtcIsac_RemoveLarMean( double* lar, WebRtc_Word16 bandwidth) argument 102 WebRtcIsac_DecorrelateIntraVec( const double* data, double* out, WebRtc_Word16 bandwidth) argument 176 WebRtcIsac_DecorrelateInterVec( const double* data, double* out, WebRtc_Word16 bandwidth) argument 245 WebRtcIsac_QuantizeUncorrLar( double* data, int* recIdx, WebRtc_Word16 bandwidth) argument 315 WebRtcIsac_DequantizeLpcParam( const int* idx, double* out, WebRtc_Word16 bandwidth) argument 371 WebRtcIsac_CorrelateIntraVec( const double* data, double* out, WebRtc_Word16 bandwidth) argument 434 WebRtcIsac_CorrelateInterVec( const double* data, double* out, WebRtc_Word16 bandwidth) argument 499 WebRtcIsac_AddLarMean( double* data, WebRtc_Word16 bandwidth) argument [all...] |
H A D | lpc_analysis.c | 305 /* bandwidth expansion */ 332 /* bandwidth expansion */ 373 * -bandwidth : specifies if the codec is in 0-16 kHz mode or 391 WebRtc_Word16 bandwidth) 396 WebRtc_Word16 numSubFrames = SUBFRAMES * (1 + (bandwidth == isac16kHz)); 442 (bandwidth == isac12kHz); 444 (bandwidth == isac16kHz); 453 /* bandwidth expansion */ 385 WebRtcIsac_GetLpcCoefUb( double* inSignal, MaskFiltstr* maskdata, double* lpCoeff, double corrMat[][UB_LPC_ORDER + 1], double* varscale, WebRtc_Word16 bandwidth) argument
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/external/chromium_org/third_party/opus/src/celt/ |
H A D | celt.h | 60 int bandwidth; member in struct:__anon13962
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/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/automated/ |
H A D | vie_network_test.cc | 155 unsigned int bandwidth = 0; local 157 &bandwidth)); 175 unsigned int bandwidth = 0; local 177 &bandwidth)); 185 unsigned int bandwidth = 0; local 187 &bandwidth)); 188 EXPECT_GT(bandwidth, 0u); 211 unsigned int bandwidth = 0; local 213 &bandwidth)); 214 EXPECT_GT(bandwidth, 222 unsigned int bandwidth = 0; local [all...] |
/external/libopus/celt/ |
H A D | celt.h | 60 int bandwidth; member in struct:__anon23727
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/external/srec/srec/cfront/ |
H A D | frontobj.c | 352 int fmax, i, j, high_cut, bandwidth; local 380 bandwidth = parameters->samplerate / 2; 381 ASSERT(bandwidth != 0); 407 fmax = bandwidth; 410 freqobj->cut_off_below = (int)(((long)freqobj->low_cut * freqobj->np) / (2.0 * bandwidth)); 411 freqobj->cut_off_above = (int)(((long)high_cut * freqobj->np) / (2.0 * bandwidth));
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/external/chromium_org/media/cast/test/utility/ |
H A D | udp_proxy.cc | 46 // Packets are output at a maximum bandwidth. 97 scoped_ptr<PacketPipe> NewBuffer(size_t buffer_size, double bandwidth) { argument 98 return scoped_ptr<PacketPipe>(new Buffer(buffer_size, bandwidth)).Pass();
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/external/chromium_org/third_party/opus/src/src/ |
H A D | analysis.c | 220 int bandwidth=0; local 382 bandwidth = 0; 414 bandwidth = b; 417 bandwidth = 20; 610 info->bandwidth = bandwidth; 611 /*printf("%d %d\n", info->bandwidth, info->opus_bandwidth);*/
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H A D | opus_decoder.c | 67 int bandwidth; member in struct:OpusDecoder 347 if( st->bandwidth == OPUS_BANDWIDTH_NARROWBAND ) { 349 } else if( st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND ) { 351 } else if( st->bandwidth == OPUS_BANDWIDTH_WIDEBAND ) { 422 switch(st->bandwidth) 650 st->bandwidth = packet_bandwidth; 670 st->bandwidth = packet_bandwidth; 792 *value = st->bandwidth; 891 int bandwidth; local 894 bandwidth [all...] |
H A D | opus_demo.c | 58 fprintf(stderr, "-bandwidth <NB|MB|WB|SWB|FB> : audio bandwidth (from narrowband to fullband); default: sampling rate\n" ); 248 int bandwidth=-1; local 339 bandwidth = OPUS_AUTO; 355 } else if( strcmp( argv[ args ], "-bandwidth" ) == 0 ) { 356 check_encoder_option(decode_only, "-bandwidth"); 358 bandwidth = OPUS_BANDWIDTH_NARROWBAND; 360 bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; 362 bandwidth = OPUS_BANDWIDTH_WIDEBAND; 364 bandwidth [all...] |