Searched defs:loss_rate (Results 1 - 17 of 17) sorted by relevance

/external/chromium_org/third_party/webrtc/voice_engine/
H A Dnetwork_predictor.cc22 void NetworkPredictor::UpdatePacketLossRate(uint8_t loss_rate) { argument
27 static_cast<float>(loss_rate));
H A Dchannel.cc1492 uint8_t loss_rate = network_predictor_->GetLossRate(); local
1494 if (audio_coding_->SetPacketLossRate(100 * loss_rate / 255) != 0) {
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
H A DPacketLossTest.cc31 int loss_rate,
33 loss_rate_ = loss_rate;
27 Setup(AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, int channels, int loss_rate, int burst_length) argument
/external/chromium_org/net/quic/congestion_control/
H A Dsend_algorithm_simulator.h126 void set_forward_loss_rate(float loss_rate) { argument
127 DCHECK_LT(loss_rate, 1.0f);
128 forward_loss_rate_ = loss_rate;
131 void set_reverse_loss_rate(float loss_rate) { argument
132 DCHECK_LT(loss_rate, 1.0f);
133 reverse_loss_rate_ = loss_rate;
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/
H A Dneteq_quality_test.cc79 // to achieve the target packet loss rate |loss_rate|, when a packet is not
82 static double ProbTrans00Solver(int units, double loss_rate, argument
85 return prob_trans_10 / (1.0f - loss_rate) - prob_trans_10;
86 // 0 == prob_trans_00 ^ (units - 1) + (1 - loss_rate) / prob_trans_10 *
87 // prob_trans_00 - (1 - loss_rate) * (1 + 1 / prob_trans_10).
99 const double a = (1.0f - loss_rate) / prob_trans_10;
100 const double b = (loss_rate - 1.0f) * (1.0f + 1.0f / prob_trans_10);
161 UniformLoss::UniformLoss(double loss_rate) argument
162 : loss_rate_(loss_rate) {
223 // pi_1 * prob_trans_10_ ^ (units - 1) == 1 - loss_rate
231 double loss_rate = 0.01f * packet_loss_rate_; local
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H A Dneteq_quality_test.h41 UniformLoss(double loss_rate);
43 void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; } argument
/external/libvpx/libvpx/examples/
H A Ddecode_with_partial_drops.c140 void throw_packets(unsigned char* frame, int* size, int loss_rate, argument
175 int loss_event = ((rand() + 1.0)/(RAND_MAX + 1.0) < loss_rate/100.0);
/external/chromium_org/third_party/opus/src/celt/
H A Dquant_bands.c264 int force_intra, opus_val32 *delayedIntra, int two_pass, int loss_rate, int lfe)
278 intra_bias = (opus_int32)((budget**delayedIntra*loss_rate)/(C*512));
261 quant_coarse_energy(const CELTMode *m, int start, int end, int effEnd, const opus_val16 *eBands, opus_val16 *oldEBands, opus_uint32 budget, opus_val16 *error, ec_enc *enc, int C, int LM, int nbAvailableBytes, int force_intra, opus_val32 *delayedIntra, int two_pass, int loss_rate, int lfe) argument
H A Dcelt_encoder.c75 int loss_rate; member in struct:OpusCustomEncoder
1052 if (st->loss_rate>2)
1054 if (st->loss_rate>4)
1056 if (st->loss_rate>8)
1719 &st->delayedIntra, st->complexity >= 4, st->loss_rate, st->lfe);
2221 st->loss_rate = value;
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/
H A Dopus_interface.c93 int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) { argument
96 OPUS_SET_PACKET_LOSS_PERC(loss_rate));
/external/chromium_org/third_party/webrtc/video_engine/test/libvietest/testbed/
H A Dtb_external_transport.cc534 bool TbExternalTransport::UniformLoss(int loss_rate) { argument
536 return (dropThis < loss_rate);
539 bool TbExternalTransport::GilbertElliotLoss(int loss_rate, int burst_length) { argument
557 double probTrans01 = (probTrans10 * ( loss_rate / (100.0 - loss_rate)));
/external/libopus/celt/
H A Dquant_bands.c264 int force_intra, opus_val32 *delayedIntra, int two_pass, int loss_rate, int lfe)
278 intra_bias = (opus_int32)((budget**delayedIntra*loss_rate)/(C*512));
261 quant_coarse_energy(const CELTMode *m, int start, int end, int effEnd, const opus_val16 *eBands, opus_val16 *oldEBands, opus_uint32 budget, opus_val16 *error, ec_enc *enc, int C, int LM, int nbAvailableBytes, int force_intra, opus_val32 *delayedIntra, int two_pass, int loss_rate, int lfe) argument
H A Dcelt_encoder.c75 int loss_rate; member in struct:OpusCustomEncoder
1052 if (st->loss_rate>2)
1054 if (st->loss_rate>4)
1056 if (st->loss_rate>8)
1719 &st->delayedIntra, st->complexity >= 4, st->loss_rate, st->lfe);
2221 st->loss_rate = value;
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testFec/
H A Dtest_packet_masks_metrics.cc260 double loss_rate = static_cast<double>( local
268 result *= (1.0 - loss_rate);
270 result *= loss_rate;
281 double prob01 = prob10 * (loss_rate / (1.0 - loss_rate));
287 result = (1.0 - loss_rate);
289 result = loss_rate;
544 float loss_rate = loss_model_[k].average_loss_rate;
548 loss_rate,
853 float loss_rate
[all...]
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A Drtp_player.cc326 float loss_rate, uint32_t rtt_ms, bool reordering)
333 loss_rate_(loss_rate),
475 const PayloadTypes& payload_types, float loss_rate, uint32_t rtt_ms,
488 payload_types, clock, &packet_source, loss_rate, rtt_ms, reordering));
323 RtpPlayerImpl(PayloadSinkFactoryInterface* payload_sink_factory, const PayloadTypes& payload_types, Clock* clock, scoped_ptr<test::RtpFileReader>* packet_source, float loss_rate, uint32_t rtt_ms, bool reordering) argument
473 Create(const std::string& input_filename, PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock, const PayloadTypes& payload_types, float loss_rate, uint32_t rtt_ms, bool reordering) argument
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/
H A Daudio_coding_module_impl.cc1480 int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) { argument
1483 codecs_[current_send_codec_idx_]->SetPacketLossRate(loss_rate) < 0) {
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Drtcp_packet.h1017 void LossRate(uint8_t loss_rate) { metric_.lossRate = loss_rate; } argument

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