| /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 *
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| /external/tlsdate/src/events/ |
| kickoff_time_sync.c | 33 add_jitter (int base, int jitter)
36 if (!jitter)
55 return base + (abs (n) % (2 * jitter)) - jitter;
91 state->opts.jitter);
123 int jitter = 0; local
128 /* Add jitter iff we had network synchronization once before. */
130 jitter = add_jitter (30, 30); /* TODO(wad) make configurable */
160 trigger_event (state, E_TLSDATE, jitter);
168 trigger_event (state, E_TLSDATE, jitter + RESOLVER_TIMEOUT)
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| /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/omx/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/avahi/avahi-common/ |
| timeval.c | 86 struct timeval *avahi_elapse_time(struct timeval *tv, unsigned msec, unsigned jitter) {
94 if (jitter) {
114 /* We use the same jitter for 10 seconds. That way our
118 avahi_timeval_add(tv, (AvahiUsec) (jitter*1000.0*r/(RAND_MAX+1.0)));
|
| /external/tlsdate/etc/ |
| tlsdated.conf | 7 jitter 0
|
| /external/strace/tests/ |
| adjtimex.c | 48 ", jitter=%jd, shift=%d, stabil=%jd, jitcnt=%jd, calcnt=%jd"
66 (intmax_t) tx.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/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/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
|
| /frameworks/base/packages/EasterEgg/src/com/android/egg/neko/ |
| NekoService.java | 117 long jitter = (long)(INTERVAL_JITTER_FRAC * interval); local
118 interval += (long)(Math.random() * (2 * jitter)) - jitter;
|
| /development/ndk/platforms/android-3/include/linux/ |
| timex.h | 50 long jitter; member in struct:timex
|
| /external/strace/ |
| print_timex.c | 61 tprintf(", tick=%jd, ppsfreq=%jd, jitter=%jd",
62 (intmax_t) tx.tick, (intmax_t) tx.ppsfreq, (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-glibc2.11-4.8/sysroot/usr/include/linux/ |
| timex.h | 80 long jitter; /* pps jitter (us) (ro) */ member in struct:timex
83 long jitcnt; /* jitter limit exceeded (ro) */
135 #define STA_PPSJITTER 0x0200 /* PPS signal jitter exceeded (ro) */
|
| /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) */
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