Searched refs:jitter (Results 1 - 25 of 60) sorted by relevance

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/external/chromium_org/third_party/webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/
H A DRtcpStatistics.java20 public final int jitter; field in class:RtcpStatistics
25 int jitter, int rttMs) {
29 this.jitter = jitter;
24 RtcpStatistics(int fractionLost, int cumulativeLost, int extendedMax, int jitter, int rttMs) argument
/external/speex/libspeex/
H A Djitter.c4 Adaptive jitter buffer for Speex
49 + jitter
67 #define SPEEX_JITTER_MAX_BUFFER_SIZE 200 /**< Maximum number of packets in jitter buffer */
169 /** Based on available data, this computes the optimal delay for the jitter buffer.
175 static spx_int16_t compute_opt_delay(JitterBuffer *jitter) argument
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
274 JitterBuffer *jitter = (JitterBuffer*)speex_alloc(sizeof(JitterBuffer)); local
297 jitter_buffer_reset(JitterBuffer *jitter) argument
328 jitter_buffer_destroy(JitterBuffer *jitter) argument
335 update_timings(JitterBuffer *jitter, spx_int32_t timing) argument
356 shift_timings(JitterBuffer *jitter, spx_int16_t amount) argument
368 jitter_buffer_put(JitterBuffer *jitter, const JitterBufferPacket *packet) argument
465 jitter_buffer_get(JitterBuffer *jitter, JitterBufferPacket *packet, spx_int32_t desired_span, spx_int32_t *start_offset) argument
680 jitter_buffer_get_another(JitterBuffer *jitter, JitterBufferPacket *packet) argument
716 _jitter_buffer_update_delay(JitterBuffer *jitter, JitterBufferPacket *packet, spx_int32_t *start_offset) argument
739 jitter_buffer_update_delay(JitterBuffer *jitter, JitterBufferPacket *packet, spx_int32_t *start_offset) argument
749 jitter_buffer_get_pointer_timestamp(JitterBuffer *jitter) argument
754 jitter_buffer_tick(JitterBuffer *jitter) argument
770 jitter_buffer_remaining_span(JitterBuffer *jitter, spx_uint32_t rem) argument
783 jitter_buffer_ctl(JitterBuffer *jitter, int request, void *ptr) argument
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/external/chromium_org/third_party/speex/include/speex/
H A Dspeex_jitter.h4 @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 buffe
[all...]
/external/speex/include/speex/
H A Dspeex_jitter.h4 @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 buffe
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/external/chromium_org/tools/oopif/
H A Diframe_server.py73 # jitter is the amount of randomness applied to nframes and nsites.
74 # Should be from [0,1]. 0.0 means no jitter.
75 # size_jitter is like jitter, but for width and height.
79 self.jitter = float(query_dict.get('jitter', [0] )[0])
116 host = rand.randint(1, apply_jitter(rand, params.jitter, params.nsites))
120 def apply_jitter(rand, jitter, n):
121 """Reduce n by random amount from [0, jitter]. Ensures result is >=1."""
122 if jitter <= 0.001:
124 v = n - int(n * rand.uniform(0, jitter))
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/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/
H A Dremote_bitrate_estimators_test.cc149 JitterFilter jitter(this);
151 jitter.SetJitter(20);
156 JitterFilter jitter(this);
158 jitter.SetJitter(i);
165 JitterFilter jitter(this);
168 jitter.SetJitter(10.0f * i);
171 jitter.SetJitter(0.0f);
229 JitterFilter jitter(this);
232 jitter.SetJitter(120);
/external/chromium_org/media/cast/net/
H A Dcast_transport_config.cc47 jitter(0),
H A Dcast_transport_config.h165 uint32 jitter; member in struct:media::cast::RtcpReportBlock
/external/chromium_org/media/cast/net/rtp/
H A Dreceiver_stats.h24 uint32* jitter) OVERRIDE;
H A Dreceiver_stats_unittest.cc36 float jitter = 0; local
41 jitter += (float_interval - jitter) / 16;
43 return static_cast<uint32>(jitter + 0.5f);
H A Dreceiver_stats.cc30 uint32* jitter) {
69 *jitter = static_cast<uint32>(std::abs(jitter_.InMillisecondsRoundedUp()));
104 // Update jitter.
27 GetStatistics(uint8* fraction_lost, uint32* cumulative_lost, uint32* extended_high_sequence_number, uint32* jitter) argument
/external/chromium_org/third_party/webrtc/video_engine/include/
H A Dvie_rtp_rtcp.h16 // - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
323 unsigned int& jitter,
332 jitter = stats.jitter;
339 unsigned int& jitter,
348 jitter = stats.jitter;
319 GetReceivedRTCPStatistics(const int video_channel, unsigned short& fraction_lost, unsigned int& cumulative_lost, unsigned int& extended_max, unsigned int& jitter, int& rtt_ms) const argument
335 GetSentRTCPStatistics(const int video_channel, unsigned short& fraction_lost, unsigned int& cumulative_lost, unsigned int& extended_max, unsigned int& jitter, int& rtt_ms) const argument
/external/chromium_org/components/invalidation/
H A Dregistration_manager_unittest.cc23 // Fake registration manager that lets you override jitter.
33 void SetJitter(double jitter) { argument
34 jitter_ = jitter;
174 void RunBackoffTest(double jitter) { argument
175 fake_registration_manager_.SetJitter(jitter);
191 jitter * RegistrationManager::kRegistrationDelayMaxJitter;
260 int GetRoundedBackoff(double retry_interval, double jitter) { argument
273 jitter,
H A Dregistration_manager.h88 // Calculate exponential backoff. |jitter| must be Uniform[-1.0, 1.0].
94 double jitter,
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/
H A Drtp_rtcp_defines.h160 extendedHighSeqNum(0), jitter(0), lastSR(0),
168 uint32_t jitter,
176 jitter(jitter),
186 uint32_t jitter; member in struct:webrtc::RTCPReportBlock
163 RTCPReportBlock(uint32_t remote_ssrc, uint32_t source_ssrc, uint8_t fraction_lost, uint32_t cumulative_lost, uint32_t extended_high_sequence_number, uint32_t jitter, uint32_t last_sender_report, uint32_t delay_since_last_sender_report) argument
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
H A Drtcp.cc45 // Calculate jitter according to RFC 3550, and update previous timestamps.
93 stats->jitter = jitter_ >> 4; // Scaling from Q4.
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A DplotReceiveTrace.m7 %DEBUG ; ( 7:59:42:500 | 0) VIDEO:-1 ; 2500; Received complete frame timestamp 1870514263 frame type 1 frame size 7862 at time 19965, jitter estimate was 130
89 [p, count] = sscanf(message, 'Received complete frame timestamp %lu frame type %u frame size %*u at time %lu, jitter estimate was %lu');
182 legend('Packet arrives', 'Frame complete', 'Decode', 'Decode complete', 'Time to render', 'Only jitter', 'Must decode');
198 % legend('Complete time - Estimated arrival time', 'Desired jitter buffer level', 'Actual decode time', 'Max decode time', 0);
H A DplotJitterEstimate.m24 plot(x, randJitters(x,2)); title('Random jitter');
/external/chromium_org/third_party/webrtc/voice_engine/
H A Dvoe_rtp_rtcp_impl.h37 unsigned int* jitter = NULL,
/external/kernel-headers/original/uapi/linux/
H A Dtimex.h80 __kernel_long_t jitter; /* pps jitter (us) (ro) */ member in struct:timex
83 __kernel_long_t jitcnt; /* jitter limit exceeded (ro) */
141 #define STA_PPSJITTER 0x0200 /* PPS signal jitter exceeded (ro) */
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Dreceive_statistics_unittest.cc156 // Add some arbitrary data, with loss and jitter.
188 EXPECT_EQ(statistics.jitter, callback.stats_.jitter);
192 EXPECT_EQ(4u, statistics.jitter);
/external/chromium_org/media/cast/net/rtcp/
H A Drtcp.h46 uint32* jitter) = 0;
H A Drtcp_unittest.cc69 uint32* jitter) OVERRIDE {
73 *jitter = 0;
/external/chromium_org/third_party/webrtc/video/
H A Dsend_statistics_proxy_unittest.cc78 EXPECT_EQ(a.rtcp_stats.jitter, b.rtcp_stats.jitter);
103 ssrc_stats.rtcp_stats.jitter = offset + 3;
117 ssrc_stats.rtcp_stats.jitter = offset + 3;
/external/iproute2/tc/
H A Dq_netem.c207 if (get_ticks(&opt.jitter, *argv)) {
440 if (dist_data && (opt.latency == 0 || opt.jitter == 0)) {
441 fprintf(stderr, "distribution specified but no latency and jitter values\n");
558 if (qopt.jitter) {
559 fprintf(f, " %s", sprint_ticks(qopt.jitter, b1));

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