| /external/chromium_org/chrome/browser/net/spdyproxy/ |
| 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_->Enable(false
[all...] |
| /external/chromium_org/net/quic/ |
| quic_config_test.cc | 26 config_.SetDefaults();
29 QuicConfig config_; member in class:net::test::__anon14075::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_.ToHandshakeMessage(&msg)
[all...] |
| quic_server_test.cc | 25 dispatcher_(config_,
38 QuicConfig config_; member in class:net::test::__anon14112::QuicChromeServerDispatchPacketTest
|
| /external/chromium_org/ppapi/cpp/ |
| audio.h | 72 AudioConfig& config() { return config_; }
79 const AudioConfig& config() const { return config_; }
92 AudioConfig config_; member in class:pp::Audio
|
| /external/chromium_org/third_party/webrtc/video/ |
| 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_.rtp.rtx.end())
[all...] |
| 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_.rtp.nack.rtp_history_ms > 0)
[all...] |
| 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()) {
|
| 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_.rtp.rtx.ssrcs.end()
[all...] |
| /external/chromium_org/components/component_updater/ |
| component_updater_ping_manager.h | 25 const Configurator& config_; member in class:component_updater::PingManager
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test/ |
| 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_,
|
| 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_.codec_settings->startBitrate)
[all...] |
| 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;
|
| 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_.codec_settings->startBitrate = start_bitrate_
[all...] |
| /external/chromium_org/media/cast/sender/ |
| 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_->g_error_resilient = 1
[all...] |
| /external/chromium_org/media/filters/ |
| 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_.is_encrypted())
[all...] |
| 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_.codec(
[all...] |
| /external/chromium_org/remoting/protocol/ |
| fake_session.cc | 15 config_(SessionConfig::ForTest()),
39 return config_;
43 config_ = config;
|
| /external/chromium_org/media/cast/net/rtp/ |
| 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/third_party/webrtc/test/ |
| 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);
|
| /external/chromium_org/content/browser/speech/ |
| 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_.audio_sample_rate
[all...] |
| /external/chromium_org/chromecast/media/cma/base/ |
| 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())
|
| /external/chromium_org/net/ssl/ |
| 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::__anon14258::MockSSLConfigService
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| 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_.open_handshake_role == InternalDataChannelInit::kOpener)
[all...] |
| /external/chromium_org/net/tools/quic/ |
| quic_server_test.cc | 24 dispatcher_(config_,
37 QuicConfig config_; member in class:net::tools::test::__anon14395::QuicServerDispatchPacketTest
|
| /external/chromium_org/content/renderer/pepper/ |
| ppb_audio_impl.cc | 50 PpapiGlobals::Get()->GetResourceTracker()->AddRefResource(config_);
51 return config_;
80 config_ = config;
118 EnterResourceNoLock<PPB_AudioConfig_API> enter(config_, true);
|
