/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | encode_lpc_swb.h | 34 * -bandwidth : indicates if the given LAR vectors belong 44 int16_t bandwidth); 56 * -bandwidth : indicates if the given LAR vectors belong 65 int16_t bandwidth); 79 * -bandwidth : indicates if the given LAR vectors belong 88 int16_t bandwidth); 98 * -bandwidth : indicates if the given LAR vectors belong 108 int16_t bandwidth); 118 * -bandwidth : indicates if the given LAR vectors belong 127 int16_t bandwidth); [all...] |
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.h | 48 int16_t bandwidth);
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H A D | entropy_coding.h | 83 int16_t bandwidth); 107 * - bandwidth : specifies if the codec is operating at 0-12 kHz 131 int16_t bandwidth, 140 * - bandwidth : spepecifies if the codec is in 0-12 kHz or 164 int16_t bandwidth); 261 * Encode if the bandwidth of encoded audio is 0-12 kHz or 0-16 kHz. 264 * - bandwidth : an enumerator specifying if the codec in is 275 int16_t WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth, 281 * Decode the bandwidth of the encoded audio, i.e. if the bandwidth i [all...] |
/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | encode_lpc_swb.h | 35 * -bandwidth : indicates if the given LAR vectors belong 45 WebRtc_Word16 bandwidth); 57 * -bandwidth : indicates if the given LAR vectors belong 66 WebRtc_Word16 bandwidth); 80 * -bandwidth : indicates if the given LAR vectors belong 89 WebRtc_Word16 bandwidth); 99 * -bandwidth : indicates if the given LAR vectors belong 109 WebRtc_Word16 bandwidth); 119 * -bandwidth : indicates if the given LAR vectors belong 128 WebRtc_Word16 bandwidth); [all...] |
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.h | 48 WebRtc_Word16 bandwidth);
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H A D | entropy_coding.h | 83 WebRtc_Word16 bandwidth); 107 * - bandwidth : specifies if the codec is operating at 0-12 kHz 131 WebRtc_Word16 bandwidth, 140 * - bandwidth : spepecifies if the codec is in 0-12 kHz or 164 WebRtc_Word16 bandwidth); 261 * Encode if the bandwidth of encoded audio is 0-12 kHz or 0-16 kHz. 264 * - bandwidth : an enumerator specifying if the codec in is 275 WebRtc_Word16 WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth, 281 * Decode the bandwidth of the encoded audio, i.e. if the bandwidth i [all...] |
/external/chromium_org/net/quic/ |
H A D | quic_sustained_bandwidth_recorder.cc | 24 QuicBandwidth bandwidth, 43 // bandwidth estimate as a valid sustained bandwidth estimate. 47 bandwidth_estimate_ = bandwidth; 48 DVLOG(1) << "New sustained bandwidth estimate (KBytes/s): " 52 // Check for an increase in max bandwidth. 53 if (bandwidth > max_bandwidth_estimate_) { 54 max_bandwidth_estimate_ = bandwidth; 56 DVLOG(1) << "New max bandwidth estimate (KBytes/s): " 22 RecordEstimate(bool in_recovery, bool in_slow_start, QuicBandwidth bandwidth, QuicTime estimate_time, QuicWallTime wall_time, QuicTime::Delta srtt) argument
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H A D | quic_sustained_bandwidth_recorder_test.cc | 23 QuicBandwidth bandwidth = local 30 recorder.RecordEstimate(in_recovery, in_slow_start, bandwidth, estimate_time, 37 recorder.RecordEstimate(in_recovery, in_slow_start, bandwidth, estimate_time, 44 recorder.RecordEstimate(in_recovery, in_slow_start, bandwidth, estimate_time, 47 EXPECT_EQ(recorder.BandwidthEstimate(), bandwidth); 56 recorder.RecordEstimate(in_recovery, in_slow_start, bandwidth, estimate_time, 59 recorder.RecordEstimate(in_recovery, in_slow_start, bandwidth, estimate_time, 61 EXPECT_EQ(recorder.BandwidthEstimate(), bandwidth); 73 // Reset again, this time recording a lower bandwidth than before. 88 // Max bandwidth shoul 103 QuicBandwidth bandwidth = local [all...] |
H A D | quic_sustained_bandwidth_recorder.h | 18 // This class keeps track of a sustained bandwidth estimate to ultimately send 19 // to the client in a server config update message. A sustained bandwidth 28 // bandwidth estimate. 29 // |time_now| is used as a max bandwidth timestamp if needed. 32 QuicBandwidth bandwidth, 64 // True if we have been able to calculate sustained bandwidth, over at least 71 // True if the current sustained bandwidth estimate was generated while in 75 // The latest sustained bandwidth estimate. 78 // The maximum sustained bandwidth seen over the lifetime of the connection.
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/external/iproute2/tc/ |
H A D | tc_red.h | 6 extern int tc_red_eval_idle_damping(int wlog, unsigned avpkt, unsigned bandwidth, __u8 *sbuf);
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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 | bandwidthsmoother.h | 19 // The purpose of BandwidthSmoother is to smooth out bandwidth 21 // are "sure" there is sufficient bandwidth. To avoid frequent fluctuations, 22 // we take a slightly pessimistic view of our bandwidth. We only increase 23 // our estimation when we have sampled bandwidth measurements of values 25 // for at least time_between_increase time. If a sampled bandwidth 28 // We retain the initial bandwidth guess as our current bandwidth estimation 39 // Samples a new bandwidth measurement. 40 // bandwidth is expected to be non-negative. 41 // returns true if the bandwidth estimatio [all...] |
/external/iproute2/testsuite/tests/ |
H A D | cbq.t | 3 $TC qdisc add dev $DEV root handle 10:0 cbq bandwidth 100Mbit avpkt 1400 mpu 64 4 $TC class add dev $DEV parent 10:0 classid 10:12 cbq bandwidth 100mbit rate 100mbit allot 1514 prio 3 maxburst 1 avpkt 500 bounded 8 $TC qdisc add dev $DEV root handle 10:0 cbq bandwidth 100Mbit avpkt 1400 mpu 64 9 $TC class add dev $DEV parent 10:0 classid 10:12 cbq bandwidth 100mbit rate 100mbit allot 1514 prio 3 maxburst 1 avpkt 500 bounded
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H A D | policer | 3 $TC qdisc add dev $DEV root handle 10:0 cbq bandwidth 100Mbit avpkt 1400 mpu 64 4 $TC class add dev $DEV parent 10:0 classid 10:12 cbq bandwidth 100mbit rate 100mbit allot 1514 prio 3 maxburst 1 avpkt 500 bounded 10 $TC qdisc add dev $DEV root handle 10:0 cbq bandwidth 100Mbit avpkt 1400 mpu 64 11 $TC class add dev $DEV parent 10:0 classid 10:12 cbq bandwidth 100mbit rate 100mbit allot 1514 prio 3 maxburst 1 avpkt 500 bounded
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H A D | cls-testbed.t | 17 cbq bandwidth 100Mbit avpkt 1400 mpu 64 20 cbq bandwidth 100mbit rate 100mbit allot 1514 prio 3 \
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/external/fio/tools/plot/samples/ |
H A D | Makefile | 1 all: clean m2sw1-128k-sdb-randwrite-para.results_bw.log io bandwidth 9 bandwidth: setup
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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/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/chromium_org/media/tools/constrained_network_server/ |
H A D | traffic_control.py | 16 # The maximum bandwidth limit. 60 Imposes packet level constraints such as bandwidth, latency, and packet loss 70 bandwidth: Maximum allowed upload bandwidth (integer in kbit/s). 106 bandwidth: Maximum allowed upload bandwidth (integer in kbit/s). 184 The class specifies bandwidth, and qdisc specifies delay and packet loss. The 192 bandwidth: Maximum allowed upload bandwidth (integer in kbit/s). 197 if 'bandwidth' no [all...] |
/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/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...] |
/external/chromium_org/third_party/webrtc/modules/bitrate_controller/include/ |
H A D | bitrate_controller.h | 11 * RTCP module. It will aggregate the results and run one bandwidth estimation 42 * RTCPBandwidthObservers). It does one aggregated send side bandwidth 58 // Gets the available payload bandwidth in bits per second. Note that 59 // this bandwidth excludes packet headers. 60 virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0; 63 * Set the start and max send bitrate used by the bandwidth management.
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/external/chromium_org/third_party/webrtc/modules/bitrate_controller/ |
H A D | send_side_bandwidth_estimation.h | 33 void UpdateReceiverEstimate(uint32_t bandwidth);
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