| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| 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...] |
| 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 : indicates if the given LAR vectors belon
[all...] |
| lpc_analysis.h | 48 int16_t bandwidth);
|
| 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 is 0-12 kH
[all...] |
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
| 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...] |
| 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 : indicates if the given LAR vectors belon
[all...] |
| lpc_analysis.h | 48 WebRtc_Word16 bandwidth);
|
| 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 is 0-12 kH
[all...] |
| /external/chromium_org/net/quic/ |
| 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): "
|
| 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 should not have changed
103 QuicBandwidth bandwidth = local
[all...] |
| 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.
|
| /external/iproute2/tc/ |
| tc_red.h | 6 extern int tc_red_eval_idle_damping(int wlog, unsigned avpkt, unsigned bandwidth, __u8 *sbuf);
|
| /external/chromium_org/third_party/webrtc/base/ |
| 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) {
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.
|
| 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 estimation change
[all...] |
| /external/iproute2/testsuite/tests/ |
| 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
|
| 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
|
| /external/chromium_org/net/base/ |
| 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
95 VLOG(1) << "Bandwidth: " << kbp
[all...] |
| /external/fio/tools/plot/samples/ |
| Makefile | 1 all: clean m2sw1-128k-sdb-randwrite-para.results_bw.log io bandwidth
9 bandwidth: setup
|
| /external/chromium_org/third_party/webrtc/video_engine/test/auto_test/automated/ |
| 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, 0u)
222 unsigned int bandwidth = 0; local
[all...] |
| /external/tcpdump/ |
| 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,
|
| /external/chromium_org/third_party/webrtc/modules/bitrate_controller/include/ |
| 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.
|
| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| psy_configuration.h | 96 Word16 bandwidth,
101 Word16 bandwidth,
|
| /system/extras/tests/memtest/ |
| Android.mk | 10 bandwidth.cpp \
|
| /external/aac/libAACenc/src/ |
| 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;
285 AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth(INT* bandWidth,
306 *bandWidth = proposedBandWidth
[all...] |
| /external/chromium_org/third_party/webrtc/modules/bitrate_controller/ |
| send_side_bandwidth_estimation.h | 33 void UpdateReceiverEstimate(uint32_t bandwidth);
|
