/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | network_predictor.cc | 22 void NetworkPredictor::UpdatePacketLossRate(uint8_t loss_rate) { argument 27 static_cast<float>(loss_rate));
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H A D | channel.cc | 1492 uint8_t loss_rate = network_predictor_->GetLossRate(); local 1494 if (audio_coding_->SetPacketLossRate(100 * loss_rate / 255) != 0) {
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | PacketLossTest.cc | 31 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
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/external/chromium_org/net/quic/congestion_control/ |
H A D | send_algorithm_simulator.h | 126 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;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | neteq_quality_test.cc | 79 // 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 [all...] |
H A D | neteq_quality_test.h | 41 UniformLoss(double loss_rate); 43 void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; } argument
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/external/libvpx/libvpx/examples/ |
H A D | decode_with_partial_drops.c | 140 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);
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/external/chromium_org/third_party/opus/src/celt/ |
H A D | quant_bands.c | 264 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
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H A D | celt_encoder.c | 75 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;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/ |
H A D | opus_interface.c | 93 int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) { argument 96 OPUS_SET_PACKET_LOSS_PERC(loss_rate));
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/external/chromium_org/third_party/webrtc/video_engine/test/libvietest/testbed/ |
H A D | tb_external_transport.cc | 534 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)));
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/external/libopus/celt/ |
H A D | quant_bands.c | 264 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
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H A D | celt_encoder.c | 75 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;
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testFec/ |
H A D | test_packet_masks_metrics.cc | 260 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 D | rtp_player.cc | 326 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | audio_coding_module_impl.cc | 1480 int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) { argument 1483 codecs_[current_send_codec_idx_]->SetPacketLossRate(loss_rate) < 0) {
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtcp_packet.h | 1017 void LossRate(uint8_t loss_rate) { metric_.lossRate = loss_rate; } argument
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