diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 0297867a6f..0246bba6fb 100644 --- a/webrtc/call/call.cc +++ b/webrtc/call/call.cc @@ -124,6 +124,15 @@ rtclog::StreamConfig CreateRtcLogStreamConfig( return rtclog_config; } +rtclog::StreamConfig CreateRtcLogStreamConfig( + const AudioReceiveStream::Config& config) { + rtclog::StreamConfig rtclog_config; + rtclog_config.remote_ssrc = config.rtp.remote_ssrc; + rtclog_config.local_ssrc = config.rtp.local_ssrc; + rtclog_config.rtp_extensions = config.rtp.extensions; + return rtclog_config; +} + } // namespace namespace internal { @@ -594,7 +603,7 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( const webrtc::AudioReceiveStream::Config& config) { TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); - event_log_->LogAudioReceiveStreamConfig(config); + event_log_->LogAudioReceiveStreamConfig(CreateRtcLogStreamConfig(config)); AudioReceiveStream* receive_stream = new AudioReceiveStream(transport_send_->packet_router(), config, config_.audio_state, event_log_); diff --git a/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h b/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h index 2bafe5bacd..4791c82731 100644 --- a/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h +++ b/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h @@ -36,7 +36,7 @@ class MockRtcEventLog : public RtcEventLog { void(const rtclog::StreamConfig& config)); MOCK_METHOD1(LogAudioReceiveStreamConfig, - void(const webrtc::AudioReceiveStream::Config& config)); + void(const rtclog::StreamConfig& config)); MOCK_METHOD1(LogAudioSendStreamConfig, void(const webrtc::AudioSendStream::Config& config)); diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc index 9bdf55c345..edaeabf85c 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log.cc +++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc @@ -64,8 +64,7 @@ class RtcEventLogImpl final : public RtcEventLog { void StopLogging() override; void LogVideoReceiveStreamConfig(const rtclog::StreamConfig& config) override; void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override; - void LogAudioReceiveStreamConfig( - const AudioReceiveStream::Config& config) override; + void LogAudioReceiveStreamConfig(const rtclog::StreamConfig& config) override; void LogAudioSendStreamConfig(const AudioSendStream::Config& config) override; void LogRtpHeader(PacketDirection direction, MediaType media_type, @@ -351,17 +350,17 @@ void RtcEventLogImpl::LogVideoSendStreamConfig( } void RtcEventLogImpl::LogAudioReceiveStreamConfig( - const AudioReceiveStream::Config& config) { + const rtclog::StreamConfig& config) { std::unique_ptr event(new rtclog::Event()); event->set_timestamp_us(rtc::TimeMicros()); event->set_type(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT); rtclog::AudioReceiveConfig* receiver_config = event->mutable_audio_receiver_config(); - receiver_config->set_remote_ssrc(config.rtp.remote_ssrc); - receiver_config->set_local_ssrc(config.rtp.local_ssrc); + receiver_config->set_remote_ssrc(config.remote_ssrc); + receiver_config->set_local_ssrc(config.local_ssrc); - for (const auto& e : config.rtp.extensions) { + for (const auto& e : config.rtp_extensions) { rtclog::RtpHeaderExtension* extension = receiver_config->add_header_extensions(); extension->set_name(e.uri); diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.h b/webrtc/logging/rtc_event_log/rtc_event_log.h index fb64e94005..8e8bd827c7 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log.h +++ b/webrtc/logging/rtc_event_log/rtc_event_log.h @@ -119,9 +119,9 @@ class RtcEventLog { // Logs configuration information for a video send stream. virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0; - // Logs configuration information for webrtc::AudioReceiveStream. + // Logs configuration information for an audio receive stream. virtual void LogAudioReceiveStreamConfig( - const webrtc::AudioReceiveStream::Config& config) = 0; + const rtclog::StreamConfig& config) = 0; // Logs configuration information for webrtc::AudioSendStream. virtual void LogAudioSendStreamConfig( @@ -202,7 +202,7 @@ class RtcEventLogNullImpl final : public RtcEventLog { const rtclog::StreamConfig& config) override {} void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {} void LogAudioReceiveStreamConfig( - const AudioReceiveStream::Config& config) override {} + const rtclog::StreamConfig& config) override {} void LogAudioSendStreamConfig( const AudioSendStream::Config& config) override {} void LogRtpHeader(PacketDirection direction, diff --git a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc index d2ee790944..da31615698 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc +++ b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc @@ -399,18 +399,18 @@ int main(int argc, char* argv[]) { } if (parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) { - webrtc::AudioReceiveStream::Config config; + webrtc::rtclog::StreamConfig config; parsed_stream.GetAudioReceiveConfig(i, &config); - global_streams.emplace_back(config.rtp.remote_ssrc, + global_streams.emplace_back(config.remote_ssrc, webrtc::MediaType::AUDIO, webrtc::kIncomingPacket); - global_streams.emplace_back(config.rtp.local_ssrc, + global_streams.emplace_back(config.local_ssrc, webrtc::MediaType::AUDIO, webrtc::kOutgoingPacket); if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming) { std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG" - << "\tssrc=" << config.rtp.remote_ssrc - << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl; + << "\tssrc=" << config.remote_ssrc + << "\tfeedback_ssrc=" << config.local_ssrc << std::endl; } } if (parsed_stream.GetEventType(i) == diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc index db7690c463..88f26a68d4 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc +++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc @@ -417,7 +417,7 @@ void ParsedRtcEventLog::GetVideoSendConfig(size_t index, void ParsedRtcEventLog::GetAudioReceiveConfig( size_t index, - AudioReceiveStream::Config* config) const { + rtclog::StreamConfig* config) const { RTC_CHECK_LT(index, GetNumberOfEvents()); const rtclog::Event& event = events_[index]; RTC_CHECK(config != nullptr); @@ -428,11 +428,11 @@ void ParsedRtcEventLog::GetAudioReceiveConfig( event.audio_receiver_config(); // Get SSRCs. RTC_CHECK(receiver_config.has_remote_ssrc()); - config->rtp.remote_ssrc = receiver_config.remote_ssrc(); + config->remote_ssrc = receiver_config.remote_ssrc(); RTC_CHECK(receiver_config.has_local_ssrc()); - config->rtp.local_ssrc = receiver_config.local_ssrc(); + config->local_ssrc = receiver_config.local_ssrc(); // Get header extensions. - GetHeaderExtensions(&config->rtp.extensions, + GetHeaderExtensions(&config->rtp_extensions, receiver_config.header_extensions()); } diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.h b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h index befc619cb1..d5aee96766 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.h +++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h @@ -122,10 +122,9 @@ class ParsedRtcEventLog { // Only the fields that are stored in the protobuf will be written. void GetVideoSendConfig(size_t index, rtclog::StreamConfig* config) const; - // Reads a config event to a (non-NULL) AudioReceiveStream::Config struct. + // Reads a config event to a (non-NULL) StreamConfig struct. // Only the fields that are stored in the protobuf will be written. - void GetAudioReceiveConfig(size_t index, - AudioReceiveStream::Config* config) const; + void GetAudioReceiveConfig(size_t index, rtclog::StreamConfig* config) const; // Reads a config event to a (non-NULL) AudioSendStream::Config struct. // Only the fields that are stored in the protobuf will be written. diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc index 4196854fd8..fea07b81ca 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc +++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc @@ -191,15 +191,15 @@ void GenerateVideoSendConfig(uint32_t extensions_bitvector, } void GenerateAudioReceiveConfig(uint32_t extensions_bitvector, - AudioReceiveStream::Config* config, + rtclog::StreamConfig* config, Random* prng) { // Add SSRCs for the stream. - config->rtp.remote_ssrc = prng->Rand(); - config->rtp.local_ssrc = prng->Rand(); + config->remote_ssrc = prng->Rand(); + config->local_ssrc = prng->Rand(); // Add header extensions. for (unsigned i = 0; i < kNumExtensions; i++) { if (extensions_bitvector & (1u << i)) { - config->rtp.extensions.push_back( + config->rtp_extensions.push_back( RtpExtension(kExtensionNames[i], prng->Rand())); } } @@ -783,7 +783,7 @@ class AudioReceiveConfigReadWriteTest : public ConfigReadWriteTest { RtcEventLogTestHelper::VerifyAudioReceiveStreamConfig(parsed_log, index, config); } - AudioReceiveStream::Config config; + rtclog::StreamConfig config; }; class AudioSendConfigReadWriteTest : public ConfigReadWriteTest { diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc index 395836106c..152f10f4d8 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc +++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc @@ -299,7 +299,7 @@ void RtcEventLogTestHelper::VerifyVideoSendStreamConfig( void RtcEventLogTestHelper::VerifyAudioReceiveStreamConfig( const ParsedRtcEventLog& parsed_log, size_t index, - const AudioReceiveStream::Config& config) { + const rtclog::StreamConfig& config) { const rtclog::Event& event = parsed_log.events_[index]; ASSERT_TRUE(IsValidBasicEvent(event)); ASSERT_EQ(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT, event.type()); @@ -307,32 +307,32 @@ void RtcEventLogTestHelper::VerifyAudioReceiveStreamConfig( event.audio_receiver_config(); // Check SSRCs. ASSERT_TRUE(receiver_config.has_remote_ssrc()); - EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc()); + EXPECT_EQ(config.remote_ssrc, receiver_config.remote_ssrc()); ASSERT_TRUE(receiver_config.has_local_ssrc()); - EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc()); + EXPECT_EQ(config.local_ssrc, receiver_config.local_ssrc()); // Check header extensions. - ASSERT_EQ(static_cast(config.rtp.extensions.size()), + ASSERT_EQ(static_cast(config.rtp_extensions.size()), receiver_config.header_extensions_size()); for (int i = 0; i < receiver_config.header_extensions_size(); i++) { ASSERT_TRUE(receiver_config.header_extensions(i).has_name()); ASSERT_TRUE(receiver_config.header_extensions(i).has_id()); const std::string& name = receiver_config.header_extensions(i).name(); int id = receiver_config.header_extensions(i).id(); - EXPECT_EQ(config.rtp.extensions[i].id, id); - EXPECT_EQ(config.rtp.extensions[i].uri, name); + EXPECT_EQ(config.rtp_extensions[i].id, id); + EXPECT_EQ(config.rtp_extensions[i].uri, name); } // Check consistency of the parser. - AudioReceiveStream::Config parsed_config; + rtclog::StreamConfig parsed_config; parsed_log.GetAudioReceiveConfig(index, &parsed_config); - EXPECT_EQ(config.rtp.remote_ssrc, parsed_config.rtp.remote_ssrc); - EXPECT_EQ(config.rtp.local_ssrc, parsed_config.rtp.local_ssrc); + EXPECT_EQ(config.remote_ssrc, parsed_config.remote_ssrc); + EXPECT_EQ(config.local_ssrc, parsed_config.local_ssrc); // Check header extensions. - EXPECT_EQ(config.rtp.extensions.size(), parsed_config.rtp.extensions.size()); - for (size_t i = 0; i < parsed_config.rtp.extensions.size(); i++) { - EXPECT_EQ(config.rtp.extensions[i].uri, - parsed_config.rtp.extensions[i].uri); - EXPECT_EQ(config.rtp.extensions[i].id, parsed_config.rtp.extensions[i].id); + EXPECT_EQ(config.rtp_extensions.size(), parsed_config.rtp_extensions.size()); + for (size_t i = 0; i < parsed_config.rtp_extensions.size(); i++) { + EXPECT_EQ(config.rtp_extensions[i].uri, + parsed_config.rtp_extensions[i].uri); + EXPECT_EQ(config.rtp_extensions[i].id, parsed_config.rtp_extensions[i].id); } } diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h index 2408ff2e6d..a23f50c3fa 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h +++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h @@ -28,7 +28,7 @@ class RtcEventLogTestHelper { static void VerifyAudioReceiveStreamConfig( const ParsedRtcEventLog& parsed_log, size_t index, - const AudioReceiveStream::Config& config); + const rtclog::StreamConfig& config); static void VerifyAudioSendStreamConfig( const ParsedRtcEventLog& parsed_log, size_t index, diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc index 513e3c1188..71f89092f9 100644 --- a/webrtc/tools/event_log_visualizer/analyzer.cc +++ b/webrtc/tools/event_log_visualizer/analyzer.cc @@ -357,10 +357,10 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) break; } case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: { - AudioReceiveStream::Config config; + rtclog::StreamConfig config; parsed_log_.GetAudioReceiveConfig(i, &config); - StreamId stream(config.rtp.remote_ssrc, kIncomingPacket); - extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions); + StreamId stream(config.remote_ssrc, kIncomingPacket); + extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions); audio_ssrcs_.insert(stream); break; } diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc index 10314eeabd..dddfe6d3b0 100644 --- a/webrtc/voice_engine/channel.cc +++ b/webrtc/voice_engine/channel.cc @@ -85,7 +85,7 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog { } void LogAudioReceiveStreamConfig( - const webrtc::AudioReceiveStream::Config& config) override { + const webrtc::rtclog::StreamConfig& config) override { rtc::CritScope lock(&crit_); if (event_log_) { event_log_->LogAudioReceiveStreamConfig(config);