| /external/speex/libspeex/ |
| jitter.c | 4 Adaptive jitter buffer for Speex
49 + jitter
67 #define SPEEX_JITTER_MAX_BUFFER_SIZE 200 /**< Maximum number of packets in jitter buffer */
137 /** Jitter buffer structure */
169 /** Based on available data, this computes the optimal delay for the jitter buffer.
175 static spx_int16_t compute_opt_delay(JitterBuffer *jitter)
190 tb = jitter->_tb;
200 if (jitter->latency_tradeoff != 0)
201 late_factor = jitter->latency_tradeoff * 100.0f / tot_count;
203 late_factor = jitter->auto_tradeoff * jitter->window_size/tot_count
274 JitterBuffer *jitter = (JitterBuffer*)speex_alloc(sizeof(JitterBuffer)); local
[all...] |
| /external/speex/include/speex/ |
| speex_jitter.h | 4 @brief Adaptive jitter buffer for Speex
38 /** @defgroup JitterBuffer JitterBuffer: Adaptive jitter buffer
39 * This is the jitter buffer that reorders UDP/RTP packets and adjusts the buffer size
50 /** Generic adaptive jitter buffer state */
53 /** Generic adaptive jitter buffer state */
66 spx_uint32_t user_data; /**< Put whatever data you like here (it's ignored by the jitter buffer) */
75 /** There was an error in the jitter buffer */
92 /** Assign a function to destroy unused packet. When setting that, the jitter
98 /** Tell the jitter buffer to only adjust the delay in multiples of the step parameter provided */
103 /** Tell the jitter buffer to only do concealment in multiples of the size parameter provided *
[all...] |
| /external/webrtc/webrtc/video/ |
| report_block_stats_unittest.cc | 26 block1_1_.jitter = 777;
31 block1_2_.jitter = 222;
36 block1_3_.jitter = 333;
42 block2_1_.jitter = 555;
47 block2_2_.jitter = 888;
64 block.jitter = stats.jitter;
87 EXPECT_EQ(0U, aggregated.jitter);
97 EXPECT_EQ(777U, aggregated.jitter);
103 EXPECT_EQ(222U, aggregated.jitter);
[all...] |
| report_block_stats.cc | 40 block.jitter = rtcp_stats.jitter;
60 aggregate.jitter += report_block->jitter;
73 aggregate.jitter = static_cast<uint32_t>(
74 (aggregate.jitter + report_blocks.size() / 2) / report_blocks.size());
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| extended_jitter_report.cc | 30 // | inter-arrival jitter |
35 // | inter-arrival jitter |
50 LOG(LS_WARNING) << "Packet is too small to contain all the jitter.";
63 bool ExtendedJitterReport::WithJitter(uint32_t jitter) {
65 LOG(LS_WARNING) << "Max inter-arrival jitter items reached.";
68 inter_arrival_jitters_.push_back(jitter);
85 for (uint32_t jitter : inter_arrival_jitters_) {
86 ByteWriter<uint32_t>::WriteBigEndian(packet + *index, jitter);
|
| extended_jitter_report.h | 35 bool WithJitter(uint32_t jitter);
38 uint32_t jitter(size_t index) const { function in class:webrtc::rtcp::ExtendedJitterReport
|
| report_block.h | 41 void WithJitter(uint32_t jitter) { jitter_ = jitter; }
51 uint32_t jitter() const { return jitter_; } function in class:webrtc::rtcp::ReportBlock
|
| extended_jitter_report_unittest.cc | 61 EXPECT_EQ(0x11121314U, parsed().jitter(0));
72 EXPECT_EQ(0x11121418U, parsed().jitter(0));
73 EXPECT_EQ(0x22242628U, parsed().jitter(1));
|
| report_block.cc | 33 // 12 | interarrival jitter |
74 ByteWriter<uint32_t>::WriteBigEndian(&buffer[12], jitter());
|
| /frameworks/av/media/libstagefright/bqhelper/tests/ |
| FrameDropper_test.cpp | 87 // return one of 1000, 0, -1000 as jitter.
101 int jitter = GetJitter(i); local
102 int64_t testTimeUs = frames[i].timeUs + jitter;
103 printf("time %lld, testTime %lld, jitter %d\n",
104 (long long)frames[i].timeUs, (long long)testTimeUs, jitter);
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
| bwe_test.h | 153 // Jitter model: Truncated gaussian.
154 // Maximum end-to-end jitter: 30ms = 2*standard_deviation.
167 jitter(listener, flow_id) {
175 jitter(listener, flow_ids) {
183 jitter.SetMaxJitter(kMaxJitterMs);
188 JitterFilter jitter; member in struct:webrtc::testing::bwe::DefaultEvaluationFilter
|
| /external/linux-kselftest/tools/testing/selftests/rcutorture/bin/ |
| kvm.sh | 52 jitter="-1"
68 echo " --jitter N [ maxsleep (us) [ maxspin (us) ] ]"
129 --jitter)
130 checkarg --jitter "(# threads [ sleep [ spin ] ])" $# "$2" '^-\{,1\}[0-9]\+\( \+[0-9]\+\)\{,2\} *$' '^error$'
131 jitter="$2"
316 -v jitter="$jitter" \
380 split(jitter, ja);
389 print "echo Build-only run, so suppressing jitter >> " rd "/log"
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimators_test.cc | 126 JitterFilter jitter(&uplink_, 0);
129 jitter.SetMaxJitter(20);
136 JitterFilter jitter(&uplink_, 0);
139 jitter.SetMaxJitter(i);
148 JitterFilter jitter(&uplink_, 0);
152 jitter.SetMaxJitter(10.0f * i);
155 jitter.SetMaxJitter(0.0f);
233 JitterFilter jitter(&uplink_, 0);
237 jitter.SetMaxJitter(120);
|
| /external/tensorflow/tensorflow/contrib/learn/python/learn/datasets/ |
| base.py | 119 def retry(initial_delay, max_delay, factor=2.0, jitter=0.25, is_retriable=None):
126 jitter: to avoid lockstep, the returned delay is multiplied by a random
127 number between (1-jitter) and (1+jitter). To add a 20% jitter, set
128 jitter = 0.2. Must be < 1.
130 max_delay * (1 + jitter).
137 if jitter >= 1:
138 raise ValueError('jitter must be < 1; was %f' % (jitter,))
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| rtp_rtcp_defines.h | 133 extendedHighSeqNum(0), jitter(0), lastSR(0),
141 uint32_t jitter,
149 jitter(jitter),
159 uint32_t jitter; member in struct:webrtc::RTCPReportBlock
|
| /external/strace/ |
| print_timex.c | 62 tprintf(", tick=%jd, ppsfreq=%jd, jitter=%jd",
63 (intmax_t) tx.tick, (intmax_t) tx.ppsfreq, (intmax_t) tx.jitter);
|
| /external/strace/tests/ |
| adjtimex.c | 65 ", ppsfreq=%jd, jitter=%jd, shift=%d, stabil=%jd, jitcnt=%jd"
78 (intmax_t) tx->jitter,
|
| /external/strace/tests-m32/ |
| adjtimex.c | 65 ", ppsfreq=%jd, jitter=%jd, shift=%d, stabil=%jd, jitcnt=%jd"
78 (intmax_t) tx->jitter,
|
| /external/strace/tests-mx32/ |
| adjtimex.c | 65 ", ppsfreq=%jd, jitter=%jd, shift=%d, stabil=%jd, jitcnt=%jd"
78 (intmax_t) tx->jitter,
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| rtcp.cc | 45 // Calculate jitter according to RFC 3550, and update previous timestamps.
93 stats->jitter = jitter_ >> 4; // Scaling from Q4.
|
| /external/webrtc/webrtc/voice_engine/ |
| voe_rtp_rtcp_impl.h | 36 unsigned int* jitter = NULL,
|
| /prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.15-4.8/sysroot/usr/include/i386-linux-gnu/bits/ |
| timex.h | 39 long int jitter; /* pps jitter (us) (ro) */ member in struct:timex
42 long int jitcnt; /* jitter limit exceeded (ro) */
95 #define STA_PPSJITTER 0x0200 /* PPS signal jitter exceeded (ro) */
|
| /prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.15-4.8/sysroot/usr/include/linux/ |
| timex.h | 80 long jitter; /* pps jitter (us) (ro) */ member in struct:timex
83 long jitcnt; /* jitter limit exceeded (ro) */
139 #define STA_PPSJITTER 0x0200 /* PPS signal jitter exceeded (ro) */
|
| /prebuilts/gcc/linux-x86/host/x86_64-linux-glibc2.15-4.8/sysroot/usr/include/x86_64-linux-gnu/bits/ |
| timex.h | 39 long int jitter; /* pps jitter (us) (ro) */ member in struct:timex
42 long int jitcnt; /* jitter limit exceeded (ro) */
95 #define STA_PPSJITTER 0x0200 /* PPS signal jitter exceeded (ro) */
|
| /compatibility/cdd/5_multimedia/ |
| 5_6_audio-latency.md | 27 * **cold output jitter**. The variability among separate measurements of cold
29 * **cold input jitter**. The variability among separate measurements of cold
47 * [SR] Minimize the cold output jitter
70 * [SR] Minimize the cold input jitter
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