Replace AudioSendStream::Config with rtclog::StreamConfig.
BUG=webrtc:7538
Review-Url: https://codereview.webrtc.org/2856063003
Cr-Original-Commit-Position: refs/heads/master@{#18224}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: f472699bbd955062dcf1413316d0810eb3e2dea9
diff --git a/call/call.cc b/call/call.cc
index 0246bba..87e41b0 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -133,6 +133,18 @@
return rtclog_config;
}
+rtclog::StreamConfig CreateRtcLogStreamConfig(
+ const AudioSendStream::Config& config) {
+ rtclog::StreamConfig rtclog_config;
+ rtclog_config.local_ssrc = config.rtp.ssrc;
+ rtclog_config.rtp_extensions = config.rtp.extensions;
+ if (config.send_codec_spec) {
+ rtclog_config.codecs.emplace_back(config.send_codec_spec->format.name,
+ config.send_codec_spec->payload_type, 0);
+ }
+ return rtclog_config;
+}
+
} // namespace
namespace internal {
@@ -549,7 +561,7 @@
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
- event_log_->LogAudioSendStreamConfig(config);
+ event_log_->LogAudioSendStreamConfig(CreateRtcLogStreamConfig(config));
AudioSendStream* send_stream = new AudioSendStream(
config, config_.audio_state, &worker_queue_, transport_send_.get(),
bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats());
diff --git a/logging/rtc_event_log/mock/mock_rtc_event_log.h b/logging/rtc_event_log/mock/mock_rtc_event_log.h
index 4791c82..cedc309 100644
--- a/logging/rtc_event_log/mock/mock_rtc_event_log.h
+++ b/logging/rtc_event_log/mock/mock_rtc_event_log.h
@@ -39,7 +39,7 @@
void(const rtclog::StreamConfig& config));
MOCK_METHOD1(LogAudioSendStreamConfig,
- void(const webrtc::AudioSendStream::Config& config));
+ void(const rtclog::StreamConfig& config));
MOCK_METHOD4(LogRtpHeader,
void(PacketDirection direction,
diff --git a/logging/rtc_event_log/rtc_event_log.cc b/logging/rtc_event_log/rtc_event_log.cc
index edaeabf..7469cf7 100644
--- a/logging/rtc_event_log/rtc_event_log.cc
+++ b/logging/rtc_event_log/rtc_event_log.cc
@@ -65,7 +65,7 @@
void LogVideoReceiveStreamConfig(const rtclog::StreamConfig& config) override;
void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override;
void LogAudioReceiveStreamConfig(const rtclog::StreamConfig& config) override;
- void LogAudioSendStreamConfig(const AudioSendStream::Config& config) override;
+ void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override;
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
@@ -370,16 +370,16 @@
}
void RtcEventLogImpl::LogAudioSendStreamConfig(
- const AudioSendStream::Config& config) {
+ const rtclog::StreamConfig& config) {
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
event->set_timestamp_us(rtc::TimeMicros());
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);
+ sender_config->set_ssrc(config.local_ssrc);
- for (const auto& e : config.rtp.extensions) {
+ for (const auto& e : config.rtp_extensions) {
rtclog::RtpHeaderExtension* extension =
sender_config->add_header_extensions();
extension->set_name(e.uri);
diff --git a/logging/rtc_event_log/rtc_event_log.h b/logging/rtc_event_log/rtc_event_log.h
index 8e8bd82..3f96556 100644
--- a/logging/rtc_event_log/rtc_event_log.h
+++ b/logging/rtc_event_log/rtc_event_log.h
@@ -123,9 +123,8 @@
virtual void LogAudioReceiveStreamConfig(
const rtclog::StreamConfig& config) = 0;
- // Logs configuration information for webrtc::AudioSendStream.
- virtual void LogAudioSendStreamConfig(
- const webrtc::AudioSendStream::Config& config) = 0;
+ // Logs configuration information for an audio send stream.
+ virtual void LogAudioSendStreamConfig(const rtclog::StreamConfig& 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.
@@ -203,8 +202,7 @@
void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {}
void LogAudioReceiveStreamConfig(
const rtclog::StreamConfig& config) override {}
- void LogAudioSendStreamConfig(
- const AudioSendStream::Config& config) override {}
+ void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {}
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index da31615..fab04c9 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/logging/rtc_event_log/rtc_event_log2text.cc
@@ -415,13 +415,13 @@
}
if (parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
- webrtc::AudioSendStream::Config config(nullptr);
+ webrtc::rtclog::StreamConfig config;
parsed_stream.GetAudioSendConfig(i, &config);
- global_streams.emplace_back(config.rtp.ssrc, webrtc::MediaType::AUDIO,
+ global_streams.emplace_back(config.local_ssrc, webrtc::MediaType::AUDIO,
webrtc::kOutgoingPacket);
if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing) {
std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
- << "\tssrc=" << config.rtp.ssrc << std::endl;
+ << "\tssrc=" << config.local_ssrc << std::endl;
}
}
if (!FLAGS_nortp &&
diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc
index 88f26a6..6194d3a 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -436,9 +436,8 @@
receiver_config.header_extensions());
}
-void ParsedRtcEventLog::GetAudioSendConfig(
- size_t index,
- AudioSendStream::Config* config) const {
+void ParsedRtcEventLog::GetAudioSendConfig(size_t index,
+ rtclog::StreamConfig* config) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
RTC_CHECK(config != nullptr);
@@ -448,9 +447,9 @@
const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
// Get SSRCs.
RTC_CHECK(sender_config.has_ssrc());
- config->rtp.ssrc = sender_config.ssrc();
+ config->local_ssrc = sender_config.ssrc();
// Get header extensions.
- GetHeaderExtensions(&config->rtp.extensions,
+ GetHeaderExtensions(&config->rtp_extensions,
sender_config.header_extensions());
}
diff --git a/logging/rtc_event_log/rtc_event_log_parser.h b/logging/rtc_event_log/rtc_event_log_parser.h
index d5aee96..966f00d 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/logging/rtc_event_log/rtc_event_log_parser.h
@@ -126,9 +126,9 @@
// Only the fields that are stored in the protobuf will be written.
void GetAudioReceiveConfig(size_t index, rtclog::StreamConfig* config) const;
- // Reads a config event to a (non-NULL) AudioSendStream::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 GetAudioSendConfig(size_t index, AudioSendStream::Config* config) const;
+ void GetAudioSendConfig(size_t index, rtclog::StreamConfig* 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
diff --git a/logging/rtc_event_log/rtc_event_log_unittest.cc b/logging/rtc_event_log/rtc_event_log_unittest.cc
index fea07b8..e655b4b 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -206,14 +206,14 @@
}
void GenerateAudioSendConfig(uint32_t extensions_bitvector,
- AudioSendStream::Config* config,
+ rtclog::StreamConfig* config,
Random* prng) {
// Add SSRC to the stream.
- config->rtp.ssrc = prng->Rand<uint32_t>();
+ config->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(
+ config->rtp_extensions.push_back(
RtpExtension(kExtensionNames[i], prng->Rand<int>()));
}
}
@@ -788,7 +788,7 @@
class AudioSendConfigReadWriteTest : public ConfigReadWriteTest {
public:
- AudioSendConfigReadWriteTest() : config(nullptr) {}
+ AudioSendConfigReadWriteTest() {}
void GenerateConfig(uint32_t extensions_bitvector) override {
GenerateAudioSendConfig(extensions_bitvector, &config, &prng);
}
@@ -800,7 +800,7 @@
RtcEventLogTestHelper::VerifyAudioSendStreamConfig(parsed_log, index,
config);
}
- AudioSendStream::Config config;
+ rtclog::StreamConfig config;
};
class VideoReceiveConfigReadWriteTest : public ConfigReadWriteTest {
diff --git a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
index 152f10f..8b4ea6b 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
@@ -339,37 +339,29 @@
void RtcEventLogTestHelper::VerifyAudioSendStreamConfig(
const ParsedRtcEventLog& parsed_log,
size_t index,
- const AudioSendStream::Config& config) {
+ const rtclog::StreamConfig& 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());
+ EXPECT_EQ(config.local_ssrc, sender_config.ssrc());
// Check header extensions.
- ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()),
+ 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);
+ 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);
+ rtclog::StreamConfig parsed_config;
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);
- }
+ VerifyStreamConfigsAreEqual(config, parsed_config);
}
void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
diff --git a/logging/rtc_event_log/rtc_event_log_unittest_helper.h b/logging/rtc_event_log/rtc_event_log_unittest_helper.h
index a23f50c..c0fc493 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest_helper.h
+++ b/logging/rtc_event_log/rtc_event_log_unittest_helper.h
@@ -29,10 +29,9 @@
const ParsedRtcEventLog& parsed_log,
size_t index,
const rtclog::StreamConfig& config);
- static void VerifyAudioSendStreamConfig(
- const ParsedRtcEventLog& parsed_log,
- size_t index,
- const AudioSendStream::Config& config);
+ static void VerifyAudioSendStreamConfig(const ParsedRtcEventLog& parsed_log,
+ size_t index,
+ const rtclog::StreamConfig& config);
static void VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
PacketDirection direction,
diff --git a/tools/event_log_visualizer/analyzer.cc b/tools/event_log_visualizer/analyzer.cc
index 71f8909..a34d855 100644
--- a/tools/event_log_visualizer/analyzer.cc
+++ b/tools/event_log_visualizer/analyzer.cc
@@ -365,10 +365,10 @@
break;
}
case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
- AudioSendStream::Config config(nullptr);
+ rtclog::StreamConfig config;
parsed_log_.GetAudioSendConfig(i, &config);
- StreamId stream(config.rtp.ssrc, kOutgoingPacket);
- extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
+ StreamId stream(config.local_ssrc, kOutgoingPacket);
+ extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions);
audio_ssrcs_.insert(stream);
break;
}
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index dddfe6d..bf7d30c 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -93,7 +93,7 @@
}
void LogAudioSendStreamConfig(
- const webrtc::AudioSendStream::Config& config) override {
+ const webrtc::rtclog::StreamConfig& config) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogAudioSendStreamConfig(config);
