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Replace AudioSendStream::Config with rtclog::StreamConfig.

Browse Source 
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2856063003
Cr-Commit-Position: refs/heads/master@{#18224}
This commit is contained in:
perkj 2017-05-22 10:12:26 -07:00 committed by Commit bot
parent ac8f52de70
commit f472699bbd
12 changed files with 49 additions and 49 deletions
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14
webrtc/call/call.cc
View File

@ -133,6 +133,18 @@ rtclog::StreamConfig CreateRtcLogStreamConfig(
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 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
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());

2
webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
View File

@ -39,7 +39,7 @@ class MockRtcEventLog : public RtcEventLog {
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,

8
webrtc/logging/rtc_event_log/rtc_event_log.cc
View File

@ -65,7 +65,7 @@ class RtcEventLogImpl final : public RtcEventLog {
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::LogAudioReceiveStreamConfig(
}
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);

8
webrtc/logging/rtc_event_log/rtc_event_log.h
View File

@ -123,9 +123,8 @@ class RtcEventLog {
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 @@ class RtcEventLogNullImpl final : public RtcEventLog {
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,

6
webrtc/logging/rtc_event_log/rtc_event_log2text.cc
View File

@ -415,13 +415,13 @@ int main(int argc, char* argv[]) {
}
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 &&

9
webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
View File

@ -436,9 +436,8 @@ void ParsedRtcEventLog::GetAudioReceiveConfig(
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 @@ void ParsedRtcEventLog::GetAudioSendConfig(
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());
}

4
webrtc/logging/rtc_event_log/rtc_event_log_parser.h
View File

@ -126,9 +126,9 @@ class ParsedRtcEventLog {
// 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

10
webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
View File

@ -206,14 +206,14 @@ void GenerateAudioReceiveConfig(uint32_t extensions_bitvector,
}
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 AudioReceiveConfigReadWriteTest : public ConfigReadWriteTest {
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 @@ class AudioSendConfigReadWriteTest : public ConfigReadWriteTest {
RtcEventLogTestHelper::VerifyAudioSendStreamConfig(parsed_log, index,
config);
}
AudioSendStream::Config config;
rtclog::StreamConfig config;
};
class VideoReceiveConfigReadWriteTest : public ConfigReadWriteTest {

22
webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
View File

@ -339,37 +339,29 @@ void RtcEventLogTestHelper::VerifyAudioReceiveStreamConfig(
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,

7
webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
View File

@ -29,10 +29,9 @@ class RtcEventLogTestHelper {
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,

6
webrtc/tools/event_log_visualizer/analyzer.cc
View File

@ -365,10 +365,10 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
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;
}

2
webrtc/voice_engine/channel.cc
View File

@ -93,7 +93,7 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog {
}
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);