/external/vboot_reference/utility/ |
D | bmpblk_utility.cc | 45 config_.config_filename.clear(); in BmpBlockUtil() 46 memset(&config_.header, '\0', BMPBLOCK_SIGNATURE_SIZE); in BmpBlockUtil() 47 config_.images_map.clear(); in BmpBlockUtil() 48 config_.screens_map.clear(); in BmpBlockUtil() 49 config_.localizations.clear(); in BmpBlockUtil() 77 config_.config_filename = filename; in load_yaml_config() 78 config_.images_map.clear(); in load_yaml_config() 79 config_.screens_map.clear(); in load_yaml_config() 80 config_.localizations.clear(); in load_yaml_config() 81 config_.locale_names.clear(); in load_yaml_config() [all …]
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/external/webrtc/webrtc/modules/video_coding/codecs/vp9/ |
D | vp9_impl.cc | 70 config_(NULL), in VP9EncoderImpl() 101 if (config_ != NULL) { in Release() 102 delete config_; in Release() 103 config_ = NULL; in Release() 135 config_->ss_target_bitrate[i] = config_->layer_target_bitrate[i] = in SetSvcRates() 136 static_cast<int>(static_cast<int64_t>(config_->rc_target_bitrate) * in SetSvcRates() 157 config_->ss_target_bitrate[i] = static_cast<unsigned int>( in SetSvcRates() 158 config_->rc_target_bitrate * rate_ratio[i] / total); in SetSvcRates() 160 config_->layer_target_bitrate[i] = config_->ss_target_bitrate[i]; in SetSvcRates() 162 config_->layer_target_bitrate[i * num_temporal_layers_] = in SetSvcRates() [all …]
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/external/webrtc/webrtc/video/ |
D | video_receive_stream.cc | 150 config_(config), in VideoReceiveStream() 154 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); in VideoReceiveStream() 157 config.rtp.transport_cc && UseSendSideBwe(config_.rtp.extensions); in VideoReceiveStream() 174 vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false, in VideoReceiveStream() 176 RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff) in VideoReceiveStream() 179 vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode); in VideoReceiveStream() 181 RTC_DCHECK(config_.rtp.remote_ssrc != 0); in VideoReceiveStream() 183 RTC_DCHECK(config_.rtp.local_ssrc != 0); in VideoReceiveStream() 184 RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc); in VideoReceiveStream() 186 vie_channel_->SetSSRC(config_.rtp.local_ssrc, kViEStreamTypeNormal, 0); in VideoReceiveStream() [all …]
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D | video_send_stream.cc | 125 config_(config), in VideoSendStream() 132 LOG(LS_INFO) << "VideoSendStream: " << config_.ToString(); in VideoSendStream() 133 RTC_DCHECK(!config_.rtp.ssrcs.empty()); in VideoSendStream() 173 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { in VideoSendStream() 174 const std::string& extension = config_.rtp.extensions[i].name; in VideoSendStream() 175 int id = config_.rtp.extensions[i].id; in VideoSendStream() 196 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; in VideoSendStream() 197 const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; in VideoSendStream() 200 config_.rtp.fec.red_payload_type, in VideoSendStream() 201 config_.rtp.fec.ulpfec_payload_type); in VideoSendStream() [all …]
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D | send_statistics_proxy_unittest.cc | 26 : fake_clock_(1234), config_(GetTestConfig()), avg_delay_ms_(0), in SendStatisticsProxyTest() 91 VideoSendStream::Config config_; member in webrtc::SendStatisticsProxyTest 101 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); in TEST_F() 102 it != config_.rtp.ssrcs.end(); in TEST_F() 115 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin(); in TEST_F() 116 it != config_.rtp.rtx.ssrcs.end(); in TEST_F() 159 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); in TEST_F() 160 it != config_.rtp.ssrcs.end(); in TEST_F() 172 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin(); in TEST_F() 173 it != config_.rtp.rtx.ssrcs.end(); in TEST_F() [all …]
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D | send_statistics_proxy.cc | 73 config_(config), in SendStatisticsProxy() 78 UpdateCodecTypeHistogram(config_.encoder_settings.payload_name); in SendStatisticsProxy() 221 if (std::find(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc) == in GetStatsEntry() 222 config_.rtp.ssrcs.end() && in GetStatsEntry() 223 std::find(config_.rtp.rtx.ssrcs.begin(), in GetStatsEntry() 224 config_.rtp.rtx.ssrcs.end(), in GetStatsEntry() 225 ssrc) == config_.rtp.rtx.ssrcs.end()) { in GetStatsEntry() 254 if (simulcast_idx >= config_.rtp.ssrcs.size()) { in OnSendEncodedImage() 256 << " >= " << config_.rtp.ssrcs.size() << ")."; in OnSendEncodedImage() 259 uint32_t ssrc = config_.rtp.ssrcs[simulcast_idx]; in OnSendEncodedImage()
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/external/webrtc/webrtc/modules/video_coding/codecs/test/ |
D | videoprocessor_integrationtest.cc | 111 webrtc::test::TestConfig config_; member in webrtc::VideoProcessorIntegrationTest 165 config_.input_filename = webrtc::test::ResourcePath("foreman_cif", "yuv"); in SetUpCodecConfig() 168 config_.output_filename = webrtc::test::TempFilename( in SetUpCodecConfig() 170 config_.frame_length_in_bytes = in SetUpCodecConfig() 172 config_.verbose = false; in SetUpCodecConfig() 174 config_.use_single_core = true; in SetUpCodecConfig() 176 config_.keyframe_interval = key_frame_interval_; in SetUpCodecConfig() 177 config_.networking_config.packet_loss_probability = packet_loss_; in SetUpCodecConfig() 180 config_.codec_settings = &codec_settings_; in SetUpCodecConfig() 181 config_.codec_settings->startBitrate = start_bitrate_; in SetUpCodecConfig() [all …]
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D | videoprocessor.cc | 53 config_(config), in VideoProcessorImpl() 79 bit_rate_factor_ = config_.codec_settings->maxFramerate * 0.001 * 8; // bits in Init() 87 last_encoder_frame_width_ = config_.codec_settings->width; in Init() 88 last_encoder_frame_height_ = config_.codec_settings->height; in Init() 112 if (!config_.use_single_core) { in Init() 116 encoder_->InitEncode(config_.codec_settings, nbr_of_cores, in Init() 117 config_.networking_config.max_payload_size_in_bytes); in Init() 123 init_result = decoder_->InitDecode(config_.codec_settings, nbr_of_cores); in Init() 130 if (config_.verbose) { in Init() 136 config_.codec_settings->startBitrate); in Init() [all …]
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D | videoprocessor_unittest.cc | 40 TestConfig config_; member in webrtc::test::VideoProcessorTest 48 config_.codec_settings = &codec_settings_; in SetUp() 49 config_.codec_settings->startBitrate = 100; in SetUp() 50 config_.codec_settings->width = 352; in SetUp() 51 config_.codec_settings->height = 288; in SetUp() 71 &packet_manipulator_mock_, config_, &stats_); in TEST_F() 83 &packet_manipulator_mock_, config_, &stats_); in TEST_F()
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D | packet_manipulator.cc | 25 config_(config), in PacketManipulatorImpl() 48 config_.packet_size_in_bytes); in ManipulatePackets() 59 } else if (RandomUniform() < config_.packet_loss_probability || in ManipulatePackets() 63 if (config_.packet_loss_mode == kBurst) { in ManipulatePackets() 65 active_burst_packets_ = config_.packet_loss_burst_length - 1; in ManipulatePackets()
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/external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
D | audio_encoder_opus.cc | 107 static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1); in MaxEncodedBytes() 118 return config_.num_channels; in NumChannels() 130 return config_.bitrate_bps; in GetTargetBitrate() 150 rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels), in EncodeInternal() 157 info.payload_type = config_.payload_type; in EncodeInternal() 164 RTC_CHECK(RecreateEncoderInstance(config_)); in Reset() 168 auto conf = config_; in SetFec() 174 auto conf = config_; in SetDtx() 180 auto conf = config_; in SetApplication() 193 auto conf = config_; in SetMaxPlaybackRate() [all …]
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D | audio_encoder_opus.h | 84 ApplicationMode application() const { return config_.application; } in application() 85 bool dtx_enabled() const { return config_.dtx_enabled; } in dtx_enabled() 92 Config config_; variable
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/external/webrtc/talk/app/webrtc/ |
D | datachannel.cc | 182 config_ = config; in Init() 184 switch (config_.open_handshake_role) { in Init() 228 return config_.maxRetransmits == -1 && in reliable() 229 config_.maxRetransmitTime == -1; in reliable() 296 ASSERT(config_.id < 0 && sid >= 0 && data_channel_type_ == cricket::DCT_SCTP); in SetSctpSid() 297 if (config_.id == sid) { in SetSctpSid() 301 config_.id = sid; in SetSctpSid() 312 if (config_.id >= 0) { in OnTransportChannelCreated() 313 provider_->AddSctpDataStream(config_.id); in OnTransportChannelCreated() 346 (data_channel_type_ == cricket::DCT_RTP) ? receive_ssrc_ : config_.id; in OnDataReceived() [all …]
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D | datachannel.h | 139 virtual bool ordered() const { return config_.ordered; } in ordered() 141 return config_.maxRetransmitTime; in maxRetransmitTime() 143 virtual uint16_t maxRetransmits() const { return config_.maxRetransmits; } in maxRetransmits() 144 virtual std::string protocol() const { return config_.protocol; } in protocol() 145 virtual bool negotiated() const { return config_.negotiated; } in negotiated() 146 virtual int id() const { return config_.id; } in id() 260 InternalDataChannelInit config_; variable
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/external/webrtc/webrtc/test/ |
D | fake_encoder.cc | 43 config_ = *config; in InitEncode() 44 target_bitrate_kbps_ = config_.startBitrate; in InitEncode() 51 assert(config_.maxFramerate > 0); in Encode() 52 int64_t time_since_last_encode_ms = 1000 / config_.maxFramerate; in Encode() 60 if (time_since_last_encode_ms > 3 * 1000 / config_.maxFramerate) { in Encode() 63 time_since_last_encode_ms = 3 * 1000 / config_.maxFramerate; in Encode() 69 config_.simulcastStream[0].minBitrate * time_since_last_encode_ms); in Encode() 78 assert(config_.numberOfSimulcastStreams > 0); in Encode() 79 for (unsigned char i = 0; i < config_.numberOfSimulcastStreams; ++i) { in Encode() 85 config_.simulcastStream[i].minBitrate * time_since_last_encode_ms); in Encode() [all …]
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D | fake_network_pipe.cc | 77 config_(config), in FakeNetworkPipe() 100 config_ = config; // Shallow copy of the struct. in SetConfig() 109 if (config_.queue_length_packets > 0 && in SendPacket() 110 capacity_link_.size() >= config_.queue_length_packets) { in SendPacket() 120 if (config_.link_capacity_kbps > 0) in SendPacket() 121 capacity_delay_ms = data_length / (config_.link_capacity_kbps / 8); in SendPacket() 166 if (UniformLoss(config_.loss_percent)) { in Process() 173 int extra_delay = GaussianRandom(config_.queue_delay_ms, in Process() 174 config_.delay_standard_deviation_ms); in Process()
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/external/webrtc/talk/media/webrtc/ |
D | fakewebrtccall.cc | 40 const webrtc::AudioSendStream::Config& config) : config_(config) { in FakeAudioSendStream() 46 return config_; in GetConfig() 73 : config_(config), received_packets_(0) { in FakeAudioReceiveStream() 79 return config_; in GetConfig() 104 config_(config), in FakeVideoSendStream() 112 return config_; in GetConfig() 183 if (config_.encoder_settings.payload_name == "VP8") { in ReconfigureVideoEncoder() 186 } else if (config_.encoder_settings.payload_name == "VP9") { in ReconfigureVideoEncoder() 191 << config_.encoder_settings.payload_name; in ReconfigureVideoEncoder() 212 : config_(config), receiving_(false) { in FakeVideoReceiveStream() [all …]
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/external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
D | audio_encoder_cng_unittest.cc | 42 config_.speech_encoder = &mock_encoder_; in AudioEncoderCngTest() 46 config_.vad = mock_vad_; in AudioEncoderCngTest() 47 config_.payload_type = kCngPayloadType; in AudioEncoderCngTest() 72 cng_.reset(new AudioEncoderCng(config_)); in CreateCng() 122 EXPECT_EQ(static_cast<size_t>(config_.num_cng_coefficients + 1), in CheckBlockGrouping() 187 AudioEncoderCng::Config config_; member in webrtc::AudioEncoderCngTest 270 if ((i % (config_.sid_frame_interval_ms / 10)) < kBlocksPerFrame) { in TEST_F() 274 EXPECT_EQ(static_cast<size_t>(config_.num_cng_coefficients) + 1, in TEST_F() 365 EXPECT_EQ(static_cast<size_t>(config_.num_cng_coefficients) + 1, in TEST_F() 387 EXPECT_EQ(static_cast<size_t>(config_.num_cng_coefficients) + 1, in TEST_F() [all …]
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/external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
D | audio_encoder_ilbc.cc | 45 : config_(config), in AudioEncoderIlbc() 127 info.payload_type = config_.payload_type; in EncodeInternal() 134 RTC_CHECK(config_.IsOk()); in Reset() 136 const int encoder_frame_size_ms = config_.frame_size_ms > 30 in Reset() 137 ? config_.frame_size_ms / 2 in Reset() 138 : config_.frame_size_ms; in Reset()
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/external/autotest/client/deps/glbench/src/ |
D | egl_stuff.cc | 23 surface_ = eglCreateWindowSurface(display_, config_, native_window, NULL); in Init() 45 if (!config_) { in GetXVisual() 70 eglChooseConfig(display_, attribs, &config_, 1, &num_configs); in GetXVisual() 79 eglGetConfigAttrib(display_, config_, EGL_NATIVE_VISUAL_ID, &visual_id); in GetXVisual() 114 CHECK(config_); in CreateContext() 115 return eglCreateContext(display_, config_, NULL, attribs); in CreateContext()
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/external/webrtc/webrtc/audio/ |
D | audio_receive_stream.cc | 87 : config_(config), in AudioReceiveStream() 90 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); in AudioReceiveStream() 91 RTC_DCHECK_NE(config_.voe_channel_id, -1); in AudioReceiveStream() 97 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); in AudioReceiveStream() 135 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); in ~AudioReceiveStream() 138 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc); in ~AudioReceiveStream() 193 stats.remote_ssrc = config_.rtp.remote_ssrc; in GetStats() 198 if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) { in GetStats() 245 return config_; in config()
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D | audio_send_stream.cc | 63 : config_(config), audio_state_(audio_state) { in AudioSendStream() 64 LOG(LS_INFO) << "AudioSendStream: " << config_.ToString(); in AudioSendStream() 65 RTC_DCHECK_NE(config_.voe_channel_id, -1); in AudioSendStream() 70 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); in AudioSendStream() 94 LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString(); in ~AudioSendStream() 128 stats.local_ssrc = config_.rtp.ssrc; in GetStats() 146 if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) { in GetStats() 210 return config_; in config()
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
D | audio_encoder_isac_t_impl.h | 110 if (config_.adaptive_mode) in GetTargetBitrate() 112 return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate; in GetTargetBitrate() 147 info.payload_type = config_.payload_type; in EncodeInternal() 153 RecreateEncoderInstance(config_); in Reset() 185 config_ = config; in RecreateEncoderInstance()
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/external/webrtc/webrtc/modules/audio_processing/intelligibility/ |
D | intelligibility_enhancer_unittest.cc | 91 config_.sample_rate_hz = kSampleRate; in IntelligibilityEnhancerTest() 92 enh_.reset(new IntelligibilityEnhancer(config_)); in IntelligibilityEnhancerTest() 96 config_.sample_rate_hz = kSampleRate; in CheckUpdate() 97 config_.var_type = step_type; in CheckUpdate() 98 enh_.reset(new IntelligibilityEnhancer(config_)); in CheckUpdate() 115 IntelligibilityEnhancer::Config config_; member in webrtc::IntelligibilityEnhancerTest
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/external/webrtc/webrtc/p2p/client/ |
D | basicportallocator.cc | 719 config_(config), in AllocationSequence() 776 if (config_ && config) { in DisableEquivalentPhases() 777 if (config_->StunServers() == config->StunServers()) { in DisableEquivalentPhases() 781 if (!config_->relays.empty()) { in DisableEquivalentPhases() 904 if (config_ && !config_->StunServers().empty()) { in CreateUDPPorts() 907 port->set_server_addresses(config_->StunServers()); in CreateUDPPorts() 946 if (!(config_ && !config_->StunServers().empty())) { in CreateStunPorts() 958 config_->StunServers(), in CreateStunPorts() 975 ASSERT(config_ && !config_->relays.empty()); in CreateRelayPorts() 976 if (!(config_ && !config_->relays.empty())) { in CreateRelayPorts() [all …]
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