Added support for keeping a buffer of the previous X seconds, to add to an AcmDump.

In addition, timestamps are now absolute instead of relative to LOG_START.

BUG=webrtc:4741
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/1209563002.

Cr-Commit-Position: refs/heads/master@{#9509}
This commit is contained in:
Ivo Creusen 2015-06-26 10:19:26 +02:00
parent 4b91bd0897
commit 241338eeb7
2 changed files with 95 additions and 69 deletions

View File

@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/main/acm2/acm_dump.h"
#include <sstream>
#include <deque>
#include "webrtc/base/checks.h"
#include "webrtc/base/thread_annotations.h"
@ -54,27 +54,35 @@ class AcmDumpImpl final : public AcmDump {
void LogDebugEvent(DebugEvent event_type) override;
private:
// Checks if the logging time has expired, and if so stops the logging.
void StopIfNecessary() EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Stops logging and clears the stored data and buffers.
void Clear() EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Returns true if the logging is currently active.
bool CurrentlyLogging() const EXCLUSIVE_LOCKS_REQUIRED(crit_) {
return active_ &&
(clock_->TimeInMicroseconds() <= start_time_us_ + duration_us_);
}
// This function is identical to LogDebugEvent, but requires holding the lock.
void LogDebugEventLocked(DebugEvent event_type,
const std::string& event_message)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Stops logging and clears the stored data and buffers.
void Clear() EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Adds a new event to the logfile if logging is active, or adds it to the
// list of recent log events otherwise.
void HandleEvent(ACMDumpEvent* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Writes the event to the file. Note that this will destroy the state of the
// input argument.
void StoreToFile(ACMDumpEvent* event) EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Adds the event to the list of recent events, and removes any events that
// are too old and no longer fall in the time window.
void AddRecentEvent(const ACMDumpEvent& event)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Amount of time in microseconds to record log events, before starting the
// actual log.
const int recent_log_duration_us = 10000000;
rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
rtc::scoped_ptr<webrtc::FileWrapper> file_ GUARDED_BY(crit_);
rtc::scoped_ptr<ACMDumpEventStream> stream_ GUARDED_BY(crit_);
bool active_ GUARDED_BY(crit_);
std::deque<ACMDumpEvent> recent_log_events_ GUARDED_BY(crit_);
bool currently_logging_ GUARDED_BY(crit_);
int64_t start_time_us_ GUARDED_BY(crit_);
int64_t duration_us_ GUARDED_BY(crit_);
const webrtc::Clock* clock_ GUARDED_BY(crit_);
const webrtc::Clock* const clock_;
};
namespace {
@ -100,7 +108,7 @@ AcmDumpImpl::AcmDumpImpl()
: crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
file_(webrtc::FileWrapper::Create()),
stream_(new webrtc::ACMDumpEventStream()),
active_(false),
currently_logging_(false),
start_time_us_(0),
duration_us_(0),
clock_(webrtc::Clock::GetRealTimeClock()) {
@ -112,38 +120,31 @@ void AcmDumpImpl::StartLogging(const std::string& file_name, int duration_ms) {
if (file_->OpenFile(file_name.c_str(), false) != 0) {
return;
}
// Add a single object to the stream that is reused at every log event.
stream_->add_stream();
active_ = true;
// Add LOG_START event to the recent event list. This call will also remove
// any events that are too old from the recent event list.
LogDebugEventLocked(DebugEvent::kLogStart, "");
currently_logging_ = true;
start_time_us_ = clock_->TimeInMicroseconds();
duration_us_ = static_cast<int64_t>(duration_ms) * 1000;
// Log the start event.
std::stringstream log_msg;
log_msg << "Initial timestamp: " << start_time_us_;
LogDebugEventLocked(DebugEvent::kLogStart, log_msg.str());
// Write all the recent events to the log file.
for (auto&& event : recent_log_events_) {
StoreToFile(&event);
}
recent_log_events_.clear();
}
void AcmDumpImpl::LogRtpPacket(bool incoming,
const uint8_t* packet,
size_t length) {
CriticalSectionScoped lock(crit_.get());
if (!CurrentlyLogging()) {
StopIfNecessary();
return;
}
// Reuse the same object at every log event.
auto rtp_event = stream_->mutable_stream(0);
rtp_event->clear_debug_event();
const int64_t timestamp = clock_->TimeInMicroseconds() - start_time_us_;
rtp_event->set_timestamp_us(timestamp);
rtp_event->set_type(webrtc::ACMDumpEvent::RTP_EVENT);
rtp_event->mutable_packet()->set_direction(
ACMDumpEvent rtp_event;
const int64_t timestamp = clock_->TimeInMicroseconds();
rtp_event.set_timestamp_us(timestamp);
rtp_event.set_type(webrtc::ACMDumpEvent::RTP_EVENT);
rtp_event.mutable_packet()->set_direction(
incoming ? ACMDumpRTPPacket::INCOMING : ACMDumpRTPPacket::OUTGOING);
rtp_event->mutable_packet()->set_rtp_data(packet, length);
std::string dump_buffer;
stream_->SerializeToString(&dump_buffer);
file_->Write(dump_buffer.data(), dump_buffer.size());
file_->Flush();
rtp_event.mutable_packet()->set_rtp_data(packet, length);
HandleEvent(&rtp_event);
}
void AcmDumpImpl::LogDebugEvent(DebugEvent event_type,
@ -157,43 +158,61 @@ void AcmDumpImpl::LogDebugEvent(DebugEvent event_type) {
LogDebugEventLocked(event_type, "");
}
void AcmDumpImpl::StopIfNecessary() {
if (active_) {
DCHECK_GT(clock_->TimeInMicroseconds(), start_time_us_ + duration_us_);
LogDebugEventLocked(DebugEvent::kLogEnd, "");
Clear();
}
void AcmDumpImpl::LogDebugEventLocked(DebugEvent event_type,
const std::string& event_message) {
ACMDumpEvent event;
int64_t timestamp = clock_->TimeInMicroseconds();
event.set_timestamp_us(timestamp);
event.set_type(webrtc::ACMDumpEvent::DEBUG_EVENT);
auto debug_event = event.mutable_debug_event();
debug_event->set_type(convertDebugEvent(event_type));
debug_event->set_message(event_message);
HandleEvent(&event);
}
void AcmDumpImpl::Clear() {
if (active_ || file_->Open()) {
if (file_->Open()) {
file_->CloseFile();
}
active_ = false;
currently_logging_ = false;
stream_->Clear();
}
void AcmDumpImpl::LogDebugEventLocked(DebugEvent event_type,
const std::string& event_message) {
if (!CurrentlyLogging()) {
StopIfNecessary();
return;
void AcmDumpImpl::HandleEvent(ACMDumpEvent* event) {
if (currently_logging_) {
if (clock_->TimeInMicroseconds() < start_time_us_ + duration_us_) {
StoreToFile(event);
} else {
LogDebugEventLocked(DebugEvent::kLogEnd, "");
Clear();
AddRecentEvent(*event);
}
} else {
AddRecentEvent(*event);
}
}
void AcmDumpImpl::StoreToFile(ACMDumpEvent* event) {
// Reuse the same object at every log event.
auto event = stream_->mutable_stream(0);
int64_t timestamp = clock_->TimeInMicroseconds() - start_time_us_;
event->set_timestamp_us(timestamp);
event->set_type(webrtc::ACMDumpEvent::DEBUG_EVENT);
event->clear_packet();
auto debug_event = event->mutable_debug_event();
debug_event->set_type(convertDebugEvent(event_type));
debug_event->set_message(event_message);
if (stream_->stream_size() < 1) {
stream_->add_stream();
}
DCHECK_EQ(stream_->stream_size(), 1);
stream_->mutable_stream(0)->Swap(event);
std::string dump_buffer;
stream_->SerializeToString(&dump_buffer);
file_->Write(dump_buffer.data(), dump_buffer.size());
}
void AcmDumpImpl::AddRecentEvent(const ACMDumpEvent& event) {
recent_log_events_.push_back(event);
while (recent_log_events_.front().timestamp_us() <
event.timestamp_us() - recent_log_duration_us) {
recent_log_events_.pop_front();
}
}
#endif // RTC_AUDIOCODING_DEBUG_DUMP
// AcmDump member functions.

View File

@ -35,12 +35,11 @@ namespace webrtc {
// back to see if they match.
class AcmDumpTest : public ::testing::Test {
public:
AcmDumpTest() : log_dumper_(AcmDump::Create()) {}
void VerifyResults(const ACMDumpEventStream& parsed_stream,
size_t packet_size) {
// Verify the result.
EXPECT_EQ(3, parsed_stream.stream_size());
const ACMDumpEvent& start_event = parsed_stream.stream(0);
EXPECT_EQ(5, parsed_stream.stream_size());
const ACMDumpEvent& start_event = parsed_stream.stream(2);
ASSERT_TRUE(start_event.has_type());
EXPECT_EQ(ACMDumpEvent::DEBUG_EVENT, start_event.type());
EXPECT_TRUE(start_event.has_timestamp_us());
@ -51,7 +50,11 @@ class AcmDumpTest : public ::testing::Test {
EXPECT_EQ(ACMDumpDebugEvent::LOG_START, start_debug_event.type());
ASSERT_TRUE(start_debug_event.has_message());
for (int i = 1; i < parsed_stream.stream_size(); i++) {
for (int i = 0; i < parsed_stream.stream_size(); i++) {
if (i == 2) {
// This is the LOG_START packet that was already verified.
continue;
}
const ACMDumpEvent& test_event = parsed_stream.stream(i);
ASSERT_TRUE(test_event.has_type());
EXPECT_EQ(ACMDumpEvent::RTP_EVENT, test_event.type());
@ -60,9 +63,9 @@ class AcmDumpTest : public ::testing::Test {
ASSERT_TRUE(test_event.has_packet());
const ACMDumpRTPPacket& test_packet = test_event.packet();
ASSERT_TRUE(test_packet.has_direction());
if (i == 1) {
if (i <= 1) {
EXPECT_EQ(ACMDumpRTPPacket::INCOMING, test_packet.direction());
} else if (i == 2) {
} else if (i >= 3) {
EXPECT_EQ(ACMDumpRTPPacket::OUTGOING, test_packet.direction());
}
ASSERT_TRUE(test_packet.has_rtp_data());
@ -88,9 +91,16 @@ class AcmDumpTest : public ::testing::Test {
const std::string temp_filename =
test::OutputPath() + test_info->test_case_name() + test_info->name();
log_dumper_->StartLogging(temp_filename, 10000000);
log_dumper_->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size());
log_dumper_->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size());
// When log_dumper goes out of scope, it causes the log file to be flushed
// to disk.
{
rtc::scoped_ptr<AcmDump> log_dumper(AcmDump::Create());
log_dumper->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size());
log_dumper->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size());
log_dumper->StartLogging(temp_filename, 10000000);
log_dumper->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size());
log_dumper->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size());
}
// Read the generated file from disk.
ACMDumpEventStream parsed_stream;
@ -102,14 +112,11 @@ class AcmDumpTest : public ::testing::Test {
// Clean up temporary file - can be pretty slow.
remove(temp_filename.c_str());
}
std::vector<uint8_t> rtp_packet_;
rtc::scoped_ptr<AcmDump> log_dumper_;
};
TEST_F(AcmDumpTest, DumpAndRead) {
Run(256, 321);
Run(256, 123);
}
} // namespace webrtc