/external/webrtc/webrtc/modules/bitrate_controller/ |
H A D | send_side_bandwidth_estimation_unittest.cc | 59 int bitrate_bps; local 62 bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms); 63 EXPECT_EQ(kInitialBitrateBps, bitrate_bps); 73 bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms); 74 EXPECT_LT(bitrate_bps, kInitialBitrateBps); 78 EXPECT_GT(bitrate_bps, kMinBitrateBps); 85 int last_bitrate_bps = bitrate_bps; 90 bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms); 92 EXPECT_EQ(last_bitrate_bps, bitrate_bps);
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/external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
H A D | audio_encoder_opus.h | 37 int bitrate_bps = 64000; member in struct:webrtc::final::Config
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/external/webrtc/webrtc/modules/pacing/ |
H A D | bitrate_prober.cc | 24 int ComputeDeltaFromBitrate(size_t packet_size, int bitrate_bps) { argument 25 assert(bitrate_bps > 0); 26 // Compute the time delta needed to send packet_size bytes at bitrate_bps 29 bitrate_bps); 55 void BitrateProber::MaybeInitializeProbe(int bitrate_bps) { argument 67 bitrates_bps[i] = kProbeBitrateMultipliers[i] * bitrate_bps;
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/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
H A D | tmmbr.cc | 55 uint32_t bitrate_bps = tmmbr_item.MaxTotalMediaBitRate * 1000; local 58 ComputeMantissaAnd6bitBase2Exponent(bitrate_bps, 17, &mantissa, &exp);
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H A D | tmmbn.cc | 53 uint32_t bitrate_bps = tmmbr_item.MaxTotalMediaBitRate * 1000; local 56 ComputeMantissaAnd6bitBase2Exponent(bitrate_bps, 17, &mantissa, &exp);
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/external/webrtc/webrtc/modules/video_coding/ |
H A D | media_optimization_unittest.cc | 33 void AddFrameAndAdvanceTime(uint32_t bitrate_bps, bool expect_frame_drop) { argument 37 size_t bytes_per_frame = bitrate_bps * frame_time_ms_ / (8 * 1000);
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
H A D | aimd_rate_control.cc | 132 void AimdRateControl::SetEstimate(int bitrate_bps, int64_t now_ms) { argument 135 current_bitrate_bps_ = ChangeBitrate(bitrate_bps, bitrate_bps, now_ms);
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H A D | remote_bitrate_estimator_unittest_helper.cc | 31 int bitrate_bps, 37 bitrate_bps_(bitrate_bps), 98 void RtpStream::set_bitrate_bps(int bitrate_bps) { argument 99 ASSERT_GE(bitrate_bps, 0); 100 bitrate_bps_ = bitrate_bps; 103 int RtpStream::bitrate_bps() const { function in class:webrtc::testing::RtpStream 139 // Divides |bitrate_bps| among all streams. The allocated bitrate per stream 141 void StreamGenerator::SetBitrateBps(int bitrate_bps) { argument 145 total_bitrate_before += it->second->bitrate_bps(); 150 bitrate_before += it->second->bitrate_bps(); 30 RtpStream(int fps, int bitrate_bps, unsigned int ssrc, unsigned int frequency, uint32_t timestamp_offset, int64_t rtcp_receive_time) argument 243 GenerateAndProcessFrame(unsigned int ssrc, unsigned int bitrate_bps) argument 285 unsigned int bitrate_bps = start_bitrate; local 312 unsigned int bitrate_bps = 0; local 406 unsigned int bitrate_bps = 30000; local 471 unsigned int bitrate_bps = SteadyStateRun(kDefaultSsrc, local [all...] |
/external/webrtc/webrtc/video/ |
H A D | receive_statistics_proxy.cc | 91 unsigned int bitrate_bps) { 94 stats_.total_bitrate_bps = bitrate_bps; 90 OnIncomingRate(unsigned int framerate, unsigned int bitrate_bps) argument
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H A D | send_statistics_proxy.cc | 244 void SendStatisticsProxy::OnSetRates(uint32_t bitrate_bps, int framerate) { argument 246 stats_.target_media_bitrate_bps = bitrate_bps; 374 stats->total_bitrate_bps = total_stats.bitrate_bps; 375 stats->retransmit_bitrate_bps = retransmit_stats.bitrate_bps;
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H A D | vie_encoder.cc | 98 virtual void OnNetworkChanged(uint32_t bitrate_bps, argument 101 owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt); 246 int bitrate_bps; local 256 bitrate_bps = last_observed_bitrate_bps_; 294 if (pad_up_to_bitrate_bps > bitrate_bps) 295 pad_up_to_bitrate_bps = bitrate_bps; 447 void ViEEncoder::OnSetRates(uint32_t bitrate_bps, int framerate) { argument 449 stats_proxy_->OnSetRates(bitrate_bps, framerate); 570 void ViEEncoder::OnNetworkChanged(uint32_t bitrate_bps, argument 573 LOG(LS_VERBOSE) << "OnNetworkChanged, bitrate" << bitrate_bps [all...] |
/external/webrtc/webrtc/voice_engine/ |
H A D | voe_codec_impl.cc | 134 int VoECodecImpl::SetBitRate(int channel, int bitrate_bps) { argument 136 "SetBitRate(bitrate_bps=%d)", bitrate_bps); 142 bitrate_bps);
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H A D | channel.cc | 1425 void Channel::SetBitRate(int bitrate_bps) { argument 1427 "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps); 1428 audio_coding_->SetBitRate(bitrate_bps);
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/external/libopus/tests/ |
H A D | test_opus_encode.c | 132 opus_int32 bitrate_bps; local 390 bitrate_bps=512000; 407 opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate_bps)); 452 bitrate_bps=((fast_rand()%508000+4000)+bitrate_bps)>>1;
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/external/webrtc/webrtc/call/ |
H A D | rampup_tests.cc | 380 void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) { argument 385 if (bitrate_bps > kExpectedHighBitrateBps) { 402 if (bitrate_bps < kExpectedLowBitrateBps && suspended) { 419 if (bitrate_bps > kExpectedHighBitrateBps && !suspended) {
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/external/libopus/src/ |
H A D | opus_demo.c | 227 opus_int32 bitrate_bps=0; local 333 bitrate_bps = (opus_int32)atol(argv[args]); 488 sweep_min = bitrate_bps; 534 opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate_bps)); 592 (long)sampling_rate, bitrate_bps*0.001, 640 while (newsize < sampling_rate/25 && bitrate_bps-fabs(sweep_bps) <= 3*12*sampling_rate/newsize) 709 bitrate_bps += sweep_bps; 712 if (bitrate_bps > sweep_max) 714 else if (bitrate_bps < sweep_min) 718 if (bitrate_bps<100 [all...] |
H A D | opus_multistream_encoder.c | 78 opus_int32 bitrate_bps; member in struct:OpusMSEncoder 420 st->bitrate_bps = OPUS_AUTO; 612 if (st->bitrate_bps > st->layout.nb_channels*40000) 615 stream_offset = st->bitrate_bps/st->layout.nb_channels/2; 627 if (st->bitrate_bps==OPUS_AUTO) 630 } else if (st->bitrate_bps==OPUS_BITRATE_MAX) 644 channel_rate = 256*(st->bitrate_bps-lfe_offset*nb_lfe-stream_offset*(nb_coupled+nb_uncoupled))/total; 718 st->variable_duration, channels, Fs, st->bitrate_bps, 760 max_data_bytes = IMIN(max_data_bytes, 3*st->bitrate_bps/(3*8*Fs/frame_size)); 775 equiv_rate = st->bitrate_bps; [all...] |
H A D | opus_encoder.c | 78 opus_int32 bitrate_bps; member in struct:OpusEncoder 222 st->bitrate_bps = 3000+Fs*channels; 822 int variable_duration, int C, opus_int32 Fs, int bitrate_bps, 833 LM = optimize_framesize(analysis_pcm, frame_size, C, Fs, bitrate_bps, 1031 st->bitrate_bps = user_bitrate_to_bitrate(st, frame_size, max_data_bytes); 1034 if (max_data_bytes<3 || st->bitrate_bps < 3*frame_rate*8 1035 || (frame_rate<50 && (max_data_bytes*frame_rate<300 || st->bitrate_bps < 2400))) 1059 cbrBytes = IMIN( (st->bitrate_bps + 4*frame_rate)/(8*frame_rate) , max_data_bytes); 1060 st->bitrate_bps = cbrBytes * (8*frame_rate); 1066 equiv_rate = st->bitrate_bps 821 compute_frame_size(const void *analysis_pcm, int frame_size, int variable_duration, int C, opus_int32 Fs, int bitrate_bps, int delay_compensation, downmix_func downmix , float *subframe_mem ) argument [all...] |
/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | rtcp_packet.h | 389 void WithBitrateBps(uint32_t bitrate_bps) { argument 390 remb_item_.BitRate = bitrate_bps;
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H A D | rtp_rtcp_impl.cc | 807 void ModuleRtpRtcpImpl::SetTargetSendBitrate(uint32_t bitrate_bps) { argument 808 rtp_sender_.SetTargetBitrate(bitrate_bps);
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | audio_coding_module_impl.cc | 256 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) { argument 260 enc->SetTargetBitrate(bitrate_bps);
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
H A D | bwe_test_framework.cc | 762 void AdaptiveVideoSource::SetBitrateBps(int bitrate_bps) { argument 763 bits_per_second_ = std::min(bitrate_bps, 2500000);
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H A D | bwe_test_framework.h | 403 virtual void SetBitrateBps(int bitrate_bps) {} argument 439 void SetBitrateBps(int bitrate_bps) override;
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/external/webrtc/webrtc/ |
H A D | common_types.h | 244 BitrateStatistics() : bitrate_bps(0), packet_rate(0), timestamp_ms(0) {} 246 uint32_t bitrate_bps; // Bitrate in bits per second. member in struct:webrtc::BitrateStatistics
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