Added logging for audio send/receive stream configs.
BUG=webrtc:4741,webrtc:6399
Review-Url: https://codereview.webrtc.org/2353543003
Cr-Commit-Position: refs/heads/master@{#14585}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 5d6bbab..9515ac1 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -370,6 +370,7 @@
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+ event_log_->LogAudioSendStreamConfig(config);
AudioSendStream* send_stream = new AudioSendStream(
config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
bitrate_allocator_.get(), event_log_);
@@ -407,6 +408,7 @@
const webrtc::AudioReceiveStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+ event_log_->LogAudioReceiveStreamConfig(config);
AudioReceiveStream* receive_stream = new AudioReceiveStream(
congestion_controller_.get(), config, config_.audio_state, event_log_);
{
diff --git a/webrtc/call/mock/mock_rtc_event_log.h b/webrtc/call/mock/mock_rtc_event_log.h
index 2762386..637389f 100644
--- a/webrtc/call/mock/mock_rtc_event_log.h
+++ b/webrtc/call/mock/mock_rtc_event_log.h
@@ -34,6 +34,12 @@
MOCK_METHOD1(LogVideoSendStreamConfig,
void(const webrtc::VideoSendStream::Config& config));
+ MOCK_METHOD1(LogAudioReceiveStreamConfig,
+ void(const webrtc::AudioReceiveStream::Config& config));
+
+ MOCK_METHOD1(LogAudioSendStreamConfig,
+ void(const webrtc::AudioSendStream::Config& config));
+
MOCK_METHOD4(LogRtpHeader,
void(PacketDirection direction,
MediaType media_type,
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc
index 00314e6..976ff23 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc
@@ -62,6 +62,9 @@
void LogVideoReceiveStreamConfig(
const VideoReceiveStream::Config& config) override;
void LogVideoSendStreamConfig(const VideoSendStream::Config& config) override;
+ void LogAudioReceiveStreamConfig(
+ const AudioReceiveStream::Config& config) override;
+ void LogAudioSendStreamConfig(const AudioSendStream::Config& config) override;
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
@@ -292,6 +295,46 @@
StoreEvent(&event);
}
+void RtcEventLogImpl::LogAudioReceiveStreamConfig(
+ const AudioReceiveStream::Config& config) {
+ std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+ event->set_timestamp_us(clock_->TimeInMicroseconds());
+ 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);
+
+ for (const auto& e : config.rtp.extensions) {
+ rtclog::RtpHeaderExtension* extension =
+ receiver_config->add_header_extensions();
+ extension->set_name(e.uri);
+ extension->set_id(e.id);
+ }
+ StoreEvent(&event);
+}
+
+void RtcEventLogImpl::LogAudioSendStreamConfig(
+ const AudioSendStream::Config& config) {
+ std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+ event->set_timestamp_us(clock_->TimeInMicroseconds());
+ event->set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
+
+ rtclog::AudioSendConfig* sender_config = event->mutable_audio_sender_config();
+
+ sender_config->set_ssrc(config.rtp.ssrc);
+
+ for (const auto& e : config.rtp.extensions) {
+ rtclog::RtpHeaderExtension* extension =
+ sender_config->add_header_extensions();
+ extension->set_name(e.uri);
+ extension->set_id(e.id);
+ }
+
+ StoreEvent(&event);
+}
+
void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.h b/webrtc/logging/rtc_event_log/rtc_event_log.h
index 910e9a6..ec57b8b 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.h
@@ -14,6 +14,8 @@
#include <memory>
#include <string>
+#include "webrtc/api/call/audio_receive_stream.h"
+#include "webrtc/api/call/audio_send_stream.h"
#include "webrtc/base/platform_file.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@@ -77,6 +79,14 @@
virtual void LogVideoSendStreamConfig(
const webrtc::VideoSendStream::Config& config) = 0;
+ // Logs configuration information for webrtc::AudioReceiveStream.
+ virtual void LogAudioReceiveStreamConfig(
+ const webrtc::AudioReceiveStream::Config& config) = 0;
+
+ // Logs configuration information for webrtc::AudioSendStream.
+ virtual void LogAudioSendStreamConfig(
+ const webrtc::AudioSendStream::Config& config) = 0;
+
// Logs the header of an incoming or outgoing RTP packet. packet_length
// is the total length of the packet, including both header and payload.
virtual void LogRtpHeader(PacketDirection direction,
@@ -123,6 +133,10 @@
const VideoReceiveStream::Config& config) override {}
void LogVideoSendStreamConfig(
const VideoSendStream::Config& config) override {}
+ void LogAudioReceiveStreamConfig(
+ const AudioReceiveStream::Config& config) override {}
+ void LogAudioSendStreamConfig(
+ const AudioSendStream::Config& config) override {}
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
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 362d79e..8f1b89d 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -104,6 +104,20 @@
return std::make_pair(varint, false);
}
+void GetHeaderExtensions(
+ std::vector<RtpExtension>* header_extensions,
+ const google::protobuf::RepeatedPtrField<rtclog::RtpHeaderExtension>&
+ proto_header_extensions) {
+ header_extensions->clear();
+ for (auto& p : proto_header_extensions) {
+ RTC_CHECK(p.has_name());
+ RTC_CHECK(p.has_id());
+ const std::string& name = p.name();
+ int id = p.id();
+ header_extensions->push_back(RtpExtension(name, id));
+ }
+}
+
} // namespace
bool ParsedRtcEventLog::ParseFile(const std::string& filename) {
@@ -311,14 +325,8 @@
config->rtp.rtx.insert(std::make_pair(map.payload_type(), rtx_pair));
}
// Get header extensions.
- config->rtp.extensions.clear();
- for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
- RTC_CHECK(receiver_config.header_extensions(i).has_name());
- RTC_CHECK(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();
- config->rtp.extensions.push_back(RtpExtension(name, id));
- }
+ GetHeaderExtensions(&config->rtp.extensions,
+ receiver_config.header_extensions());
// Get decoders.
config->decoders.clear();
for (int i = 0; i < receiver_config.decoders_size(); i++) {
@@ -347,14 +355,8 @@
config->rtp.ssrcs.push_back(sender_config.ssrcs(i));
}
// Get header extensions.
- config->rtp.extensions.clear();
- for (int i = 0; i < sender_config.header_extensions_size(); i++) {
- RTC_CHECK(sender_config.header_extensions(i).has_name());
- RTC_CHECK(sender_config.header_extensions(i).has_id());
- const std::string& name = sender_config.header_extensions(i).name();
- int id = sender_config.header_extensions(i).id();
- config->rtp.extensions.push_back(RtpExtension(name, id));
- }
+ GetHeaderExtensions(&config->rtp.extensions,
+ sender_config.header_extensions());
// Get RTX settings.
config->rtp.rtx.ssrcs.clear();
for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
@@ -376,6 +378,45 @@
sender_config.encoder().payload_type();
}
+void ParsedRtcEventLog::GetAudioReceiveConfig(
+ size_t index,
+ AudioReceiveStream::Config* config) const {
+ RTC_CHECK_LT(index, GetNumberOfEvents());
+ const rtclog::Event& event = events_[index];
+ RTC_CHECK(config != nullptr);
+ RTC_CHECK(event.has_type());
+ RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
+ RTC_CHECK(event.has_audio_receiver_config());
+ const rtclog::AudioReceiveConfig& receiver_config =
+ event.audio_receiver_config();
+ // Get SSRCs.
+ RTC_CHECK(receiver_config.has_remote_ssrc());
+ config->rtp.remote_ssrc = receiver_config.remote_ssrc();
+ RTC_CHECK(receiver_config.has_local_ssrc());
+ config->rtp.local_ssrc = receiver_config.local_ssrc();
+ // Get header extensions.
+ GetHeaderExtensions(&config->rtp.extensions,
+ receiver_config.header_extensions());
+}
+
+void ParsedRtcEventLog::GetAudioSendConfig(
+ size_t index,
+ AudioSendStream::Config* config) const {
+ RTC_CHECK_LT(index, GetNumberOfEvents());
+ const rtclog::Event& event = events_[index];
+ RTC_CHECK(config != nullptr);
+ RTC_CHECK(event.has_type());
+ RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
+ RTC_CHECK(event.has_audio_sender_config());
+ const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
+ // Get SSRCs.
+ RTC_CHECK(sender_config.has_ssrc());
+ config->rtp.ssrc = sender_config.ssrc();
+ // Get header extensions.
+ GetHeaderExtensions(&config->rtp.extensions,
+ sender_config.header_extensions());
+}
+
void ParsedRtcEventLog::GetAudioPlayout(size_t index, uint32_t* ssrc) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
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 6a684cb..2d66b90 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
@@ -99,6 +99,15 @@
// Only the fields that are stored in the protobuf will be written.
void GetVideoSendConfig(size_t index, VideoSendStream::Config* config) const;
+ // Reads a config event to a (non-NULL) AudioReceiveStream::Config struct.
+ // Only the fields that are stored in the protobuf will be written.
+ void GetAudioReceiveConfig(size_t index,
+ AudioReceiveStream::Config* 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.
+ void GetAudioSendConfig(size_t index, AudioSendStream::Config* config) const;
+
// Reads the SSRC from the audio playout event at |index|. The SSRC is stored
// in the output parameter ssrc. The output parameter can be set to nullptr
// and in that case the function only asserts that the event is well formed.
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 d6af3e9..5ca885c 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -200,6 +200,35 @@
}
}
+void GenerateAudioReceiveConfig(uint32_t extensions_bitvector,
+ AudioReceiveStream::Config* config,
+ Random* prng) {
+ // Add SSRCs for the stream.
+ config->rtp.remote_ssrc = prng->Rand<uint32_t>();
+ config->rtp.local_ssrc = prng->Rand<uint32_t>();
+ // Add header extensions.
+ for (unsigned i = 0; i < kNumExtensions; i++) {
+ if (extensions_bitvector & (1u << i)) {
+ config->rtp.extensions.push_back(
+ RtpExtension(kExtensionNames[i], prng->Rand<int>()));
+ }
+ }
+}
+
+void GenerateAudioSendConfig(uint32_t extensions_bitvector,
+ AudioSendStream::Config* config,
+ Random* prng) {
+ // Add SSRC to the stream.
+ config->rtp.ssrc = prng->Rand<uint32_t>();
+ // Add header extensions.
+ for (unsigned i = 0; i < kNumExtensions; i++) {
+ if (extensions_bitvector & (1u << i)) {
+ config->rtp.extensions.push_back(
+ RtpExtension(kExtensionNames[i], prng->Rand<int>()));
+ }
+ }
+}
+
// Test for the RtcEventLog class. Dumps some RTP packets and other events
// to disk, then reads them back to see if they match.
void LogSessionAndReadBack(size_t rtp_count,
@@ -324,9 +353,10 @@
PrintExpectedEvents(rtp_count, rtcp_count, playout_count, bwe_loss_count);
}
RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
- RtcEventLogTestHelper::VerifyReceiveStreamConfig(parsed_log, 1,
- receiver_config);
- RtcEventLogTestHelper::VerifySendStreamConfig(parsed_log, 2, sender_config);
+ RtcEventLogTestHelper::VerifyVideoReceiveStreamConfig(parsed_log, 1,
+ receiver_config);
+ RtcEventLogTestHelper::VerifyVideoSendStreamConfig(parsed_log, 2,
+ sender_config);
size_t event_index = config_count + 1;
size_t rtcp_index = 1;
size_t playout_index = 1;
@@ -457,4 +487,138 @@
remove(temp_filename.c_str());
}
+class ConfigReadWriteTest {
+ public:
+ ConfigReadWriteTest() : prng(987654321) {}
+ virtual ~ConfigReadWriteTest() {}
+ virtual void GenerateConfig(uint32_t extensions_bitvector) = 0;
+ virtual void VerifyConfig(const ParsedRtcEventLog& parsed_log,
+ size_t index) = 0;
+ virtual void LogConfig(RtcEventLog* event_log) = 0;
+
+ void DoTest() {
+ // Find the name of the current test, in order to use it as a temporary
+ // filename.
+ auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
+ const std::string temp_filename =
+ test::OutputPath() + test_info->test_case_name() + test_info->name();
+
+ // Use all extensions.
+ uint32_t extensions_bitvector = (1u << kNumExtensions) - 1;
+ GenerateConfig(extensions_bitvector);
+
+ // Log a single config event and stop logging.
+ SimulatedClock fake_clock(prng.Rand<uint32_t>());
+ std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create(&fake_clock));
+
+ log_dumper->StartLogging(temp_filename, 10000000);
+ LogConfig(log_dumper.get());
+ log_dumper->StopLogging();
+
+ // Read the generated file from disk.
+ ParsedRtcEventLog parsed_log;
+ ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
+
+ // Check the generated number of events.
+ EXPECT_EQ(3u, parsed_log.GetNumberOfEvents());
+
+ RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
+
+ // Verify that the parsed config struct matches the one that was logged.
+ VerifyConfig(parsed_log, 1);
+
+ RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 2);
+
+ // Clean up temporary file - can be pretty slow.
+ remove(temp_filename.c_str());
+ }
+ Random prng;
+};
+
+class AudioReceiveConfigReadWriteTest : public ConfigReadWriteTest {
+ public:
+ void GenerateConfig(uint32_t extensions_bitvector) override {
+ GenerateAudioReceiveConfig(extensions_bitvector, &config, &prng);
+ }
+ void LogConfig(RtcEventLog* event_log) override {
+ event_log->LogAudioReceiveStreamConfig(config);
+ }
+ void VerifyConfig(const ParsedRtcEventLog& parsed_log,
+ size_t index) override {
+ RtcEventLogTestHelper::VerifyAudioReceiveStreamConfig(parsed_log, index,
+ config);
+ }
+ AudioReceiveStream::Config config;
+};
+
+class AudioSendConfigReadWriteTest : public ConfigReadWriteTest {
+ public:
+ AudioSendConfigReadWriteTest() : config(nullptr) {}
+ void GenerateConfig(uint32_t extensions_bitvector) override {
+ GenerateAudioSendConfig(extensions_bitvector, &config, &prng);
+ }
+ void LogConfig(RtcEventLog* event_log) override {
+ event_log->LogAudioSendStreamConfig(config);
+ }
+ void VerifyConfig(const ParsedRtcEventLog& parsed_log,
+ size_t index) override {
+ RtcEventLogTestHelper::VerifyAudioSendStreamConfig(parsed_log, index,
+ config);
+ }
+ AudioSendStream::Config config;
+};
+
+class VideoReceiveConfigReadWriteTest : public ConfigReadWriteTest {
+ public:
+ VideoReceiveConfigReadWriteTest() : config(nullptr) {}
+ void GenerateConfig(uint32_t extensions_bitvector) override {
+ GenerateVideoReceiveConfig(extensions_bitvector, &config, &prng);
+ }
+ void LogConfig(RtcEventLog* event_log) override {
+ event_log->LogVideoReceiveStreamConfig(config);
+ }
+ void VerifyConfig(const ParsedRtcEventLog& parsed_log,
+ size_t index) override {
+ RtcEventLogTestHelper::VerifyVideoReceiveStreamConfig(parsed_log, index,
+ config);
+ }
+ VideoReceiveStream::Config config;
+};
+
+class VideoSendConfigReadWriteTest : public ConfigReadWriteTest {
+ public:
+ VideoSendConfigReadWriteTest() : config(nullptr) {}
+ void GenerateConfig(uint32_t extensions_bitvector) override {
+ GenerateVideoSendConfig(extensions_bitvector, &config, &prng);
+ }
+ void LogConfig(RtcEventLog* event_log) override {
+ event_log->LogVideoSendStreamConfig(config);
+ }
+ void VerifyConfig(const ParsedRtcEventLog& parsed_log,
+ size_t index) override {
+ RtcEventLogTestHelper::VerifyVideoSendStreamConfig(parsed_log, index,
+ config);
+ }
+ VideoSendStream::Config config;
+};
+
+TEST(RtcEventLogTest, LogAudioReceiveConfig) {
+ AudioReceiveConfigReadWriteTest test;
+ test.DoTest();
+}
+
+TEST(RtcEventLogTest, LogAudioSendConfig) {
+ AudioSendConfigReadWriteTest test;
+ test.DoTest();
+}
+
+TEST(RtcEventLogTest, LogVideoReceiveConfig) {
+ VideoReceiveConfigReadWriteTest test;
+ test.DoTest();
+}
+
+TEST(RtcEventLogTest, LogVideoSendConfig) {
+ VideoSendConfigReadWriteTest test;
+ test.DoTest();
+}
} // namespace webrtc
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 b640301..88bc9ba 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
@@ -102,7 +102,7 @@
return ::testing::AssertionSuccess();
}
-void RtcEventLogTestHelper::VerifyReceiveStreamConfig(
+void RtcEventLogTestHelper::VerifyVideoReceiveStreamConfig(
const ParsedRtcEventLog& parsed_log,
size_t index,
const VideoReceiveStream::Config& config) {
@@ -198,7 +198,7 @@
}
}
-void RtcEventLogTestHelper::VerifySendStreamConfig(
+void RtcEventLogTestHelper::VerifyVideoSendStreamConfig(
const ParsedRtcEventLog& parsed_log,
size_t index,
const VideoSendStream::Config& config) {
@@ -270,6 +270,82 @@
parsed_config.encoder_settings.payload_type);
}
+void RtcEventLogTestHelper::VerifyAudioReceiveStreamConfig(
+ const ParsedRtcEventLog& parsed_log,
+ size_t index,
+ const AudioReceiveStream::Config& config) {
+ const rtclog::Event& event = parsed_log.events_[index];
+ ASSERT_TRUE(IsValidBasicEvent(event));
+ ASSERT_EQ(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT, event.type());
+ const rtclog::AudioReceiveConfig& receiver_config =
+ event.audio_receiver_config();
+ // Check SSRCs.
+ ASSERT_TRUE(receiver_config.has_remote_ssrc());
+ EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
+ ASSERT_TRUE(receiver_config.has_local_ssrc());
+ EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
+ // Check header extensions.
+ ASSERT_EQ(static_cast<int>(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);
+ }
+
+ // Check consistency of the parser.
+ AudioReceiveStream::Config 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);
+ // 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);
+ }
+}
+
+void RtcEventLogTestHelper::VerifyAudioSendStreamConfig(
+ const ParsedRtcEventLog& parsed_log,
+ size_t index,
+ const AudioSendStream::Config& config) {
+ const rtclog::Event& event = parsed_log.events_[index];
+ ASSERT_TRUE(IsValidBasicEvent(event));
+ ASSERT_EQ(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT, event.type());
+ const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
+ // Check SSRCs.
+ EXPECT_EQ(config.rtp.ssrc, sender_config.ssrc());
+ // Check header extensions.
+ ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
+ sender_config.header_extensions_size());
+ for (int i = 0; i < sender_config.header_extensions_size(); i++) {
+ ASSERT_TRUE(sender_config.header_extensions(i).has_name());
+ ASSERT_TRUE(sender_config.header_extensions(i).has_id());
+ const std::string& name = sender_config.header_extensions(i).name();
+ int id = sender_config.header_extensions(i).id();
+ EXPECT_EQ(config.rtp.extensions[i].id, id);
+ EXPECT_EQ(config.rtp.extensions[i].uri, name);
+ }
+
+ // Check consistency of the parser.
+ AudioSendStream::Config parsed_config(nullptr);
+ parsed_log.GetAudioSendConfig(index, &parsed_config);
+ // Check SSRCs
+ EXPECT_EQ(config.rtp.ssrc, parsed_config.rtp.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);
+ }
+}
+
void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
PacketDirection direction,
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 5ffb6f4..01ade07 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
@@ -18,13 +18,22 @@
class RtcEventLogTestHelper {
public:
- static void VerifyReceiveStreamConfig(
+ static void VerifyVideoReceiveStreamConfig(
const ParsedRtcEventLog& parsed_log,
size_t index,
const VideoReceiveStream::Config& config);
- static void VerifySendStreamConfig(const ParsedRtcEventLog& parsed_log,
- size_t index,
- const VideoSendStream::Config& config);
+ static void VerifyVideoSendStreamConfig(
+ const ParsedRtcEventLog& parsed_log,
+ size_t index,
+ const VideoSendStream::Config& config);
+ static void VerifyAudioReceiveStreamConfig(
+ const ParsedRtcEventLog& parsed_log,
+ size_t index,
+ const AudioReceiveStream::Config& config);
+ static void VerifyAudioSendStreamConfig(
+ const ParsedRtcEventLog& parsed_log,
+ size_t index,
+ const AudioSendStream::Config& config);
static void VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
PacketDirection direction,
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
index 4ec1a29..cc6425f 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -353,12 +353,20 @@
}
case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
AudioReceiveStream::Config config;
- // TODO(terelius): Parse the audio configs once we have them.
+ parsed_log_.GetAudioReceiveConfig(i, &config);
+ StreamId stream(config.rtp.remote_ssrc, kIncomingPacket);
+ RegisterHeaderExtensions(config.rtp.extensions,
+ &extension_maps[stream]);
+ audio_ssrcs_.insert(stream);
break;
}
case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
AudioSendStream::Config config(nullptr);
- // TODO(terelius): Parse the audio configs once we have them.
+ parsed_log_.GetAudioSendConfig(i, &config);
+ StreamId stream(config.rtp.ssrc, kOutgoingPacket);
+ RegisterHeaderExtensions(config.rtp.extensions,
+ &extension_maps[stream]);
+ audio_ssrcs_.insert(stream);
break;
}
case ParsedRtcEventLog::RTP_EVENT: {
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index ce9770e..8f196c6 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -94,6 +94,22 @@
}
}
+ void LogAudioReceiveStreamConfig(
+ const webrtc::AudioReceiveStream::Config& config) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogAudioReceiveStreamConfig(config);
+ }
+ }
+
+ void LogAudioSendStreamConfig(
+ const webrtc::AudioSendStream::Config& config) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogAudioSendStreamConfig(config);
+ }
+ }
+
void LogRtpHeader(webrtc::PacketDirection direction,
webrtc::MediaType media_type,
const uint8_t* header,