/external/chromium_org/third_party/webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/ |
H A D | RtcpStatistics.java | 20 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
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/external/speex/libspeex/ |
H A D | 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 */ 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 [all...] |
/external/chromium_org/third_party/speex/include/speex/ |
H A D | 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 buffe [all...] |
/external/speex/include/speex/ |
H A D | 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 buffe [all...] |
/external/chromium_org/tools/oopif/ |
H A D | iframe_server.py | 73 # 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)) [all...] |
/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
H A D | remote_bitrate_estimators_test.cc | 149 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);
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/external/chromium_org/media/cast/net/ |
H A D | cast_transport_config.cc | 47 jitter(0),
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H A D | cast_transport_config.h | 165 uint32 jitter; member in struct:media::cast::RtcpReportBlock
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/external/chromium_org/media/cast/net/rtp/ |
H A D | receiver_stats.h | 24 uint32* jitter) OVERRIDE;
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H A D | receiver_stats_unittest.cc | 36 float jitter = 0; local 41 jitter += (float_interval - jitter) / 16; 43 return static_cast<uint32>(jitter + 0.5f);
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H A D | receiver_stats.cc | 30 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
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/external/chromium_org/third_party/webrtc/video_engine/include/ |
H A D | vie_rtp_rtcp.h | 16 // - 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
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/external/chromium_org/components/invalidation/ |
H A D | registration_manager_unittest.cc | 23 // 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,
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H A D | registration_manager.h | 88 // Calculate exponential backoff. |jitter| must be Uniform[-1.0, 1.0]. 94 double jitter,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
H A D | rtp_rtcp_defines.h | 160 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | rtcp.cc | 45 // Calculate jitter according to RFC 3550, and update previous timestamps. 93 stats->jitter = jitter_ >> 4; // Scaling from Q4.
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | plotReceiveTrace.m | 7 %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);
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H A D | plotJitterEstimate.m | 24 plot(x, randJitters(x,2)); title('Random jitter');
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | voe_rtp_rtcp_impl.h | 37 unsigned int* jitter = NULL,
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/external/kernel-headers/original/uapi/linux/ |
H A D | timex.h | 80 __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) */
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | receive_statistics_unittest.cc | 156 // Add some arbitrary data, with loss and jitter. 188 EXPECT_EQ(statistics.jitter, callback.stats_.jitter); 192 EXPECT_EQ(4u, statistics.jitter);
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/external/chromium_org/media/cast/net/rtcp/ |
H A D | rtcp.h | 46 uint32* jitter) = 0;
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H A D | rtcp_unittest.cc | 69 uint32* jitter) OVERRIDE { 73 *jitter = 0;
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/external/chromium_org/third_party/webrtc/video/ |
H A D | send_statistics_proxy_unittest.cc | 78 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;
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/external/iproute2/tc/ |
H A D | q_netem.c | 207 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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