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    Searched refs:sender_config (Results 1 - 5 of 5) sorted by null

  /external/webrtc/webrtc/call/
rtc_event_log_unittest.cc 187 const rtclog::VideoSendConfig& sender_config = event.video_sender_config(); local
190 sender_config.ssrcs_size());
191 for (int i = 0; i < sender_config.ssrcs_size(); i++) {
192 EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i));
196 sender_config.header_extensions_size());
197 for (int i = 0; i < sender_config.header_extensions_size(); i++) {
198 ASSERT_TRUE(sender_config.header_extensions(i).has_name());
199 ASSERT_TRUE(sender_config.header_extensions(i).has_id());
200 const std::string& name = sender_config.header_extensions(i).name();
201 int id = sender_config.header_extensions(i).id()
    [all...]
rtc_event_log.cc 306 rtclog::VideoSendConfig* sender_config = event.mutable_video_sender_config();
309 sender_config->add_ssrcs(ssrc);
314 sender_config->add_header_extensions();
320 sender_config->add_rtx_ssrcs(rtx_ssrc);
322 sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type);
324 rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
call_perf_tests.cc 244 Call::Config sender_config; local
245 sender_config.audio_state = AudioState::Create(send_audio_state_config);
247 receiver_config.audio_state = sender_config.audio_state;
248 CreateCalls(sender_config, receiver_config);
  /external/webrtc/webrtc/test/
call_test.h 63 void CreateCalls(const Call::Config& sender_config,
call_test.cc 159 void CallTest::CreateCalls(const Call::Config& sender_config,
161 CreateSenderCall(sender_config);

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