Add UTC time to init event in AEC debug dump.
Bug: webrtc:9616
Change-Id: I1350212f0b8835fb64427483269da96d51670c01
Reviewed-on: https://webrtc-review.googlesource.com/92620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24267}diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 9678b25..71fbcf6 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -442,6 +442,7 @@
":audioproc_test_utils",
":audioproc_unittest_proto",
"../../api/audio:audio_frame_api",
+ "../../rtc_base:rtc_base_tests_utils",
"../../rtc_base:rtc_task_queue",
"aec_dump",
"aec_dump:aec_dump_unittests",
diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.cc b/modules/audio_processing/aec_dump/aec_dump_impl.cc
index 1b8df40..d94822d 100644
--- a/modules/audio_processing/aec_dump/aec_dump_impl.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_impl.cc
@@ -74,7 +74,8 @@
thread_sync_event.Wait(rtc::Event::kForever);
}
-void AecDumpImpl::WriteInitMessage(const ProcessingConfig& api_format) {
+void AecDumpImpl::WriteInitMessage(const ProcessingConfig& api_format,
+ int64_t time_now_ms) {
auto task = CreateWriteToFileTask();
auto* event = task->GetEvent();
event->set_type(audioproc::Event::INIT);
@@ -95,6 +96,7 @@
static_cast<int32_t>(api_format.reverse_input_stream().num_channels()));
msg->set_num_reverse_output_channels(
api_format.reverse_output_stream().num_channels());
+ msg->set_timestamp_ms(time_now_ms);
worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(std::move(task)));
}
diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.h b/modules/audio_processing/aec_dump/aec_dump_impl.h
index 1ad4fc8..a5416a0 100644
--- a/modules/audio_processing/aec_dump/aec_dump_impl.h
+++ b/modules/audio_processing/aec_dump/aec_dump_impl.h
@@ -52,8 +52,8 @@
~AecDumpImpl() override;
- void WriteInitMessage(const ProcessingConfig& api_format) override;
-
+ void WriteInitMessage(const ProcessingConfig& api_format,
+ int64_t time_now_ms) override;
void AddCaptureStreamInput(const AudioFrameView<const float>& src) override;
void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override;
void AddCaptureStreamInput(const AudioFrame& frame) override;
diff --git a/modules/audio_processing/aec_dump/aec_dump_integration_test.cc b/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
index 53e731a..b6d8756 100644
--- a/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
@@ -33,7 +33,7 @@
auto mock_aec_dump =
absl::make_unique<testing::StrictMock<webrtc::test::MockAecDump>>();
EXPECT_CALL(*mock_aec_dump.get(), WriteConfig(_)).Times(AtLeast(1));
- EXPECT_CALL(*mock_aec_dump.get(), WriteInitMessage(_)).Times(AtLeast(1));
+ EXPECT_CALL(*mock_aec_dump.get(), WriteInitMessage(_, _)).Times(AtLeast(1));
return std::unique_ptr<webrtc::test::MockAecDump>(std::move(mock_aec_dump));
}
diff --git a/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/modules/audio_processing/aec_dump/aec_dump_unittest.cc
index dea149d..75ed529 100644
--- a/modules/audio_processing/aec_dump/aec_dump_unittest.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_unittest.cc
@@ -40,7 +40,8 @@
aec_dump->WriteConfig(apm_config);
webrtc::ProcessingConfig api_format;
- aec_dump->WriteInitMessage(api_format);
+ constexpr int64_t kTimeNowMs = 123456789ll;
+ aec_dump->WriteInitMessage(api_format, kTimeNowMs);
}
// Remove file after the AecDump d-tor has finished.
ASSERT_EQ(0, remove(filename.c_str()));
diff --git a/modules/audio_processing/aec_dump/mock_aec_dump.h b/modules/audio_processing/aec_dump/mock_aec_dump.h
index 95f912f..c01de51 100644
--- a/modules/audio_processing/aec_dump/mock_aec_dump.h
+++ b/modules/audio_processing/aec_dump/mock_aec_dump.h
@@ -25,7 +25,8 @@
MockAecDump();
virtual ~MockAecDump();
- MOCK_METHOD1(WriteInitMessage, void(const ProcessingConfig& api_format));
+ MOCK_METHOD2(WriteInitMessage,
+ void(const ProcessingConfig& api_format, int64_t time_now_ms));
MOCK_METHOD1(AddCaptureStreamInput,
void(const AudioFrameView<const float>& src));
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 4865afa..3711375 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -34,6 +34,7 @@
#include "rtc_base/platform_file.h"
#include "rtc_base/refcountedobject.h"
#include "rtc_base/system/arch.h"
+#include "rtc_base/timeutils.h"
#include "rtc_base/trace_event.h"
#if WEBRTC_INTELLIGIBILITY_ENHANCER
#include "modules/audio_processing/intelligibility/intelligibility_enhancer.h"
@@ -572,7 +573,7 @@
InitializePreProcessor();
if (aec_dump_) {
- aec_dump_->WriteInitMessage(formats_.api_format);
+ aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis());
}
return kNoError;
}
@@ -1575,7 +1576,7 @@
// 'aec_dump' parameter, which is after locks are released.
aec_dump_.swap(aec_dump);
WriteAecDumpConfigMessage(true);
- aec_dump_->WriteInitMessage(formats_.api_format);
+ aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis());
}
void AudioProcessingImpl::DetachAecDump() {
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 2b5501e..3776474 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -28,6 +28,7 @@
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
+#include "rtc_base/fakeclock.h"
#include "rtc_base/gtest_prod_util.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -1730,6 +1731,7 @@
}
void ApmTest::VerifyDebugDumpTest(Format format) {
+ rtc::ScopedFakeClock fake_clock;
const std::string in_filename = test::ResourcePath("ref03", "aecdump");
std::string format_string;
switch (format) {
diff --git a/modules/audio_processing/debug.proto b/modules/audio_processing/debug.proto
index f7f8d10..ebfb84d 100644
--- a/modules/audio_processing/debug.proto
+++ b/modules/audio_processing/debug.proto
@@ -14,6 +14,7 @@
optional int32 output_sample_rate = 7;
optional int32 reverse_output_sample_rate = 8;
optional int32 num_reverse_output_channels = 9;
+ optional int64 timestamp_ms = 10;
}
// May contain interleaved or deinterleaved data, but don't store both formats.
diff --git a/modules/audio_processing/include/aec_dump.h b/modules/audio_processing/include/aec_dump.h
index 25c08b8..95c010b 100644
--- a/modules/audio_processing/include/aec_dump.h
+++ b/modules/audio_processing/include/aec_dump.h
@@ -76,7 +76,11 @@
virtual ~AecDump() = default;
// Logs Event::Type INIT message.
- virtual void WriteInitMessage(const ProcessingConfig& api_format) = 0;
+ virtual void WriteInitMessage(const ProcessingConfig& api_format,
+ int64_t time_now_ms) = 0;
+ RTC_DEPRECATED void WriteInitMessage(const ProcessingConfig& api_format) {
+ WriteInitMessage(api_format, 0);
+ }
// Logs Event::Type STREAM message. To log an input/output pair,
// call the AddCapture* and AddAudioProcessingState methods followed
diff --git a/pc/statscollector.cc b/pc/statscollector.cc
index e42244a..9a77a21 100644
--- a/pc/statscollector.cc
+++ b/pc/statscollector.cc
@@ -446,8 +446,7 @@
// Wallclock time in ms.
double StatsCollector::GetTimeNow() {
- return rtc::TimeUTCMicros() /
- static_cast<double>(rtc::kNumMicrosecsPerMillisec);
+ return static_cast<double>(rtc::TimeUTCMillis());
}
// Adds a MediaStream with tracks that can be used as a |selector| in a call
diff --git a/rtc_base/timeutils.cc b/rtc_base/timeutils.cc
index f65bd15..de8fc32 100644
--- a/rtc_base/timeutils.cc
+++ b/rtc_base/timeutils.cc
@@ -201,6 +201,9 @@
}
int64_t TimeUTCMicros() {
+ if (g_clock) {
+ return g_clock->TimeNanos() / kNumNanosecsPerMicrosec;
+ }
#if defined(WEBRTC_POSIX)
struct timeval time;
gettimeofday(&time, nullptr);
@@ -217,4 +220,8 @@
#endif
}
+int64_t TimeUTCMillis() {
+ return TimeUTCMicros() / kNumMicrosecsPerMillisec;
+}
+
} // namespace rtc
diff --git a/rtc_base/timeutils.h b/rtc_base/timeutils.h
index 48b5c8d..20b1dac 100644
--- a/rtc_base/timeutils.h
+++ b/rtc_base/timeutils.h
@@ -126,6 +126,10 @@
// measuring time intervals and timeouts.
int64_t TimeUTCMicros();
+// Return the number of milliseconds since January 1, 1970, UTC.
+// See above.
+int64_t TimeUTCMillis();
+
// Interval of time from the range [min, max] inclusive.
class IntervalRange {
public:
diff --git a/rtc_tools/unpack_aecdump/unpack.cc b/rtc_tools/unpack_aecdump/unpack.cc
index 3ce6906..44f8346 100644
--- a/rtc_tools/unpack_aecdump/unpack.cc
+++ b/rtc_tools/unpack_aecdump/unpack.cc
@@ -322,6 +322,11 @@
num_reverse_channels = msg.num_reverse_channels();
fprintf(settings_file, " Reverse channels: %" PRIuS "\n",
num_reverse_channels);
+ if (msg.has_timestamp_ms()) {
+ const int64_t timestamp = msg.timestamp_ms();
+ fprintf(settings_file, " Timestamp in millisecond: %" PRId64 "\n",
+ timestamp);
+ }
fprintf(settings_file, "\n");