/external/chromium_org/chrome/browser/net/spdyproxy/ |
H A D | data_reduction_proxy_chrome_configurator_unittest.cc | 23 config_.reset(new DataReductionProxyChromeConfigurator( 46 scoped_ptr<DataReductionProxyChromeConfigurator> config_; member in class:DataReductionProxyConfigTest 51 config_->Enable(false, 63 config_->Enable(false, 76 config_->AddHostPatternToBypass("<local>"); 77 config_->AddHostPatternToBypass("*.goo.com"); 78 config_->Enable(false, 90 config_->Enable(false, false, "https://www.foo.com:443/", "", ""); 97 config_->Enable(true, 108 config_ [all...] |
/external/chromium_org/net/quic/ |
H A D | quic_config_test.cc | 26 config_.SetDefaults(); 29 QuicConfig config_; member in class:net::test::__anon9450::QuicConfigTest 33 config_.SetDefaults(); 34 config_.SetInitialFlowControlWindowToSend( 36 config_.SetInitialStreamFlowControlWindowToSend( 38 config_.SetInitialSessionFlowControlWindowToSend( 40 config_.set_idle_connection_state_lifetime(QuicTime::Delta::FromSeconds(5), 42 config_.set_max_streams_per_connection(4, 2); 43 config_.SetSocketReceiveBufferToSend(kDefaultSocketReceiveBuffer); 45 config_ [all...] |
H A D | quic_server_test.cc | 25 dispatcher_(config_, 38 QuicConfig config_; member in class:net::test::__anon9487::QuicChromeServerDispatchPacketTest
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/external/chromium_org/ppapi/cpp/ |
H A D | audio.h | 72 AudioConfig& config() { return config_; } 79 const AudioConfig& config() const { return config_; } 92 AudioConfig config_; member in class:pp::Audio
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/external/chromium_org/third_party/webrtc/video/ |
H A D | video_receive_stream.cc | 42 config_(config), 53 rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0); 55 SetRtcpMode(config_.rtp.rtcp_mode); 57 assert(config_.rtp.remote_ssrc != 0); 59 assert(config_.rtp.local_ssrc != 0); 60 assert(config_.rtp.remote_ssrc != config_.rtp.local_ssrc); 62 rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.local_ssrc); 64 Config::Rtp::RtxMap::const_iterator it = config_.rtp.rtx.begin(); 65 if (it != config_ [all...] |
H A D | video_send_stream.cc | 123 config_(config), 137 assert(config_.rtp.ssrcs.size() > 0); 139 assert(config_.rtp.min_transmit_bitrate_bps >= 0); 141 config_.rtp.min_transmit_bitrate_bps / 1000); 143 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { 144 const std::string& extension = config_.rtp.extensions[i].name; 145 int id = config_.rtp.extensions[i].id; 160 if (config_.rtp.fec.red_payload_type != -1) { 161 assert(config_.rtp.fec.ulpfec_payload_type != -1); 162 if (config_ [all...] |
H A D | send_statistics_proxy.cc | 21 : config_(config), 55 if (std::find(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc) == 56 config_.rtp.ssrcs.end() && 57 std::find(config_.rtp.rtx.ssrcs.begin(), 58 config_.rtp.rtx.ssrcs.end(), 59 ssrc) == config_.rtp.rtx.ssrcs.end()) {
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H A D | send_statistics_proxy_unittest.cc | 31 config_ = GetTestConfig(); 83 VideoSendStream::Config config_; member in class:webrtc::SendStatisticsProxyTest 92 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); 93 it != config_.rtp.ssrcs.end(); 106 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin(); 107 it != config_.rtp.rtx.ssrcs.end(); 154 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); 155 it != config_.rtp.ssrcs.end(); 166 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin(); 167 it != config_ [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test/ |
H A D | videoprocessor_unittest.cc | 40 TestConfig config_; member in class:webrtc::test::VideoProcessorTest 48 config_.codec_settings = &codec_settings_; 49 config_.codec_settings->startBitrate = 100; 50 config_.codec_settings->width = 352; 51 config_.codec_settings->height = 288; 76 &packet_manipulator_mock_, config_, 92 &packet_manipulator_mock_, config_,
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H A D | videoprocessor.cc | 53 config_(config), 78 bit_rate_factor_ = config_.codec_settings->maxFramerate * 0.001 * 8; // bits 86 last_encoder_frame_width_ = config_.codec_settings->width; 87 last_encoder_frame_height_ = config_.codec_settings->height; 107 if (!config_.use_single_core) { 111 encoder_->InitEncode(config_.codec_settings, nbr_of_cores, 112 config_.networking_config.max_payload_size_in_bytes); 118 init_result = decoder_->InitDecode(config_.codec_settings, nbr_of_cores); 125 if (config_.verbose) { 131 config_ [all...] |
H A D | packet_manipulator.cc | 23 config_(config), 47 config_.packet_size_in_bytes); 58 } else if (RandomUniform() < config_.packet_loss_probability || 62 if (config_.packet_loss_mode == kBurst) { 64 active_burst_packets_ = config_.packet_loss_burst_length - 1;
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H A D | videoprocessor_integrationtest.cc | 110 webrtc::test::TestConfig config_; member in class:webrtc::VideoProcessorIntegrationTest 157 config_.input_filename = 161 config_.output_filename = webrtc::test::TempFilename( 163 config_.frame_length_in_bytes = CalcBufferSize(kI420, 165 config_.verbose = false; 167 config_.use_single_core = true; 169 config_.keyframe_interval = key_frame_interval_; 170 config_.networking_config.packet_loss_probability = packet_loss_; 174 config_.codec_settings = &codec_settings_; 175 config_ [all...] |
/external/chromium_org/media/cast/sender/ |
H A D | vp8_encoder.cc | 53 config_.reset(new vpx_codec_enc_cfg_t()); 72 if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(), config_.get(), 0)) { 75 config_->g_w = cast_config_.width; 76 config_->g_h = cast_config_.height; 77 config_->rc_target_bitrate = cast_config_.start_bitrate / 1000; // In kbit/s. 80 config_->g_timebase.num = 1; 81 config_->g_timebase.den = kVideoFrequency; 82 config_->g_lag_in_frames = 0; 83 config_->kf_mode = VPX_KF_DISABLED; 87 config_ [all...] |
/external/chromium_org/media/filters/ |
H A D | opus_audio_decoder.cc | 262 config_ = config; 327 discard_helper_->Reset(config_.codec_delay()); 345 if (config_.codec() != kCodecOpus) { 351 ChannelLayoutToChannelCount(config_.channel_layout()); 352 if (!config_.IsValidConfig() || channel_count > kMaxVorbisChannels) { 354 << " codec: " << config_.codec() 356 << " channel layout: " << config_.channel_layout() 357 << " bits per channel: " << config_.bits_per_channel() 358 << " samples per second: " << config_.samples_per_second(); 362 if (config_ [all...] |
H A D | ffmpeg_audio_decoder.cc | 158 config_ = config; 283 if (av_frame_->sample_rate != config_.samples_per_second() || 284 channels != ChannelLayoutToChannelCount(config_.channel_layout()) || 288 << config_.samples_per_second() 290 << ChannelLayoutToChannelCount(config_.channel_layout()) 294 if (config_.codec() == kCodecAAC && 295 av_frame_->sample_rate == 2 * config_.samples_per_second()) { 310 DCHECK_EQ(ChannelLayoutToChannelCount(config_.channel_layout()), 338 if (!config_.IsValidConfig()) { 340 << " codec: " << config_ [all...] |
/external/chromium_org/remoting/protocol/ |
H A D | fake_session.cc | 15 config_(SessionConfig::ForTest()), 39 return config_; 43 config_ = config;
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/external/chromium_org/media/cast/net/rtp/ |
H A D | rtp_packetizer_unittest.cc | 34 : config_(config), 51 EXPECT_EQ(config_.ssrc, rtp_header.ssrc); 89 RtpPacketizerConfig config_; member in class:media::cast::TestRtpPacketTransport 106 config_.sequence_number = kSeqNum; 107 config_.ssrc = kSsrc; 108 config_.payload_type = kPayload; 109 config_.max_payload_length = kMaxPacketLength; 110 transport_.reset(new TestRtpPacketTransport(config_)); 117 pacer_->RegisterVideoSsrc(config_.ssrc); 119 pacer_.get(), &packet_storage_, config_)); 140 RtpPacketizerConfig config_; member in class:media::cast::RtpPacketizerTest [all...] |
/external/chromium_org/components/component_updater/ |
H A D | component_updater_ping_manager.h | 25 const Configurator& config_; member in class:component_updater::PingManager
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/external/chromium_org/third_party/webrtc/test/ |
H A D | fake_network_pipe.cc | 79 config_(config), 103 config_ = config; // Shallow copy of the struct. 112 if (config_.queue_length_packets > 0 && 113 capacity_link_.size() >= config_.queue_length_packets) { 123 if (config_.link_capacity_kbps > 0) 124 capacity_delay_ms = data_length / (config_.link_capacity_kbps / 8); 168 if (UniformLoss(config_.loss_percent)) { 175 int extra_delay = GaussianRandom(config_.queue_delay_ms, 176 config_.delay_standard_deviation_ms);
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/external/chromium_org/content/browser/speech/ |
H A D | google_one_shot_remote_engine.cc | 160 config_ = config; 166 std::string lang_param = config_.language; 192 if (!config_.grammars.empty()) { 193 DCHECK_EQ(config_.grammars.size(), 1U); 194 parts.push_back("lm=" + net::EscapeQueryParamValue(config_.grammars[0].url, 198 if (!config_.hardware_info.empty()) 199 parts.push_back("xhw=" + net::EscapeQueryParamValue(config_.hardware_info, 201 parts.push_back("maxresults=" + base::UintToString(config_.max_hypotheses)); 202 parts.push_back(config_.filter_profanities ? "pfilter=2" : "pfilter=0"); 210 config_ [all...] |
/external/chromium_org/chromecast/media/cma/base/ |
H A D | buffering_state.cc | 30 : config_(config), 91 if (buffer_duration < config_->high_level()) 100 << " low_level_ms=" << config_->low_level().InMilliseconds() 101 << " high_level_ms=" << config_->high_level().InMilliseconds(); 112 if (buffer_duration < config_->low_level()) 114 if (buffer_duration >= config_->high_level())
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/external/chromium_org/net/ssl/ |
H A D | ssl_config_service_unittest.cc | 19 explicit MockSSLConfigService(const SSLConfig& config) : config_(config) {} 23 *config = config_; 29 SSLConfig old_config = config_; 30 config_ = config; 31 ProcessConfigUpdate(old_config, config_); 37 SSLConfig config_; member in class:net::__anon9633::MockSSLConfigService
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
H A D | datachannel.cc | 142 config_ = config; 176 return config_.maxRetransmits == -1 && 177 config_.maxRetransmitTime == -1; 244 ASSERT(config_.id < 0 && sid >= 0 && data_channel_type_ == cricket::DCT_SCTP); 245 if (config_.id == sid) 248 config_.id = sid; 259 if (config_.id >= 0) { 260 provider_->AddSctpDataStream(config_.id); 293 (data_channel_type_ == cricket::DCT_RTP) ? receive_ssrc_ : config_.id; 360 if (config_ [all...] |
/external/chromium_org/net/tools/quic/ |
H A D | quic_server_test.cc | 24 dispatcher_(config_, 37 QuicConfig config_; member in class:net::tools::test::__anon9770::QuicServerDispatchPacketTest
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/external/chromium_org/content/renderer/pepper/ |
H A D | ppb_audio_impl.cc | 50 PpapiGlobals::Get()->GetResourceTracker()->AddRefResource(config_); 51 return config_; 80 config_ = config; 118 EnterResourceNoLock<PPB_AudioConfig_API> enter(config_, true);
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