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rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/webrtc/call/rtc_event_log.cc

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/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/call/rtc_event_log.h"
#include <limits>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/event.h"
#include "webrtc/base/swap_queue.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/call.h"
#include "webrtc/call/rtc_event_log_helper_thread.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
#ifdef ENABLE_RTC_EVENT_LOG
// Files generated at build-time by the protobuf compiler.
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
#else
#include "webrtc/call/rtc_event_log.pb.h"
#endif
#endif
namespace webrtc {
#ifndef ENABLE_RTC_EVENT_LOG
// No-op implementation if flag is not set.
class RtcEventLogNullImpl final : public RtcEventLog {
public:
bool StartLogging(const std::string& file_name,
int64_t max_size_bytes) override {
return false;
}
bool StartLogging(rtc::PlatformFile platform_file,
int64_t max_size_bytes) override {
// The platform_file is open and needs to be closed.
if (!rtc::ClosePlatformFile(platform_file)) {
LOG(LS_ERROR) << "Can't close file.";
}
return false;
}
void StopLogging() override {}
void LogVideoReceiveStreamConfig(
const VideoReceiveStream::Config& config) override {}
void LogVideoSendStreamConfig(
const VideoSendStream::Config& config) override {}
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length) override {}
void LogRtcpPacket(PacketDirection direction,
MediaType media_type,
const uint8_t* packet,
size_t length) override {}
void LogAudioPlayout(uint32_t ssrc) override {}
void LogBwePacketLossEvent(int32_t bitrate,
uint8_t fraction_loss,
int32_t total_packets) override {}
};
#else // ENABLE_RTC_EVENT_LOG is defined
class RtcEventLogImpl final : public RtcEventLog {
public:
explicit RtcEventLogImpl(const Clock* clock);
~RtcEventLogImpl() override;
bool StartLogging(const std::string& file_name,
int64_t max_size_bytes) override;
bool StartLogging(rtc::PlatformFile platform_file,
int64_t max_size_bytes) override;
void StopLogging() override;
void LogVideoReceiveStreamConfig(
const VideoReceiveStream::Config& config) override;
void LogVideoSendStreamConfig(const VideoSendStream::Config& config) override;
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length) override;
void LogRtcpPacket(PacketDirection direction,
MediaType media_type,
const uint8_t* packet,
size_t length) override;
void LogAudioPlayout(uint32_t ssrc) override;
void LogBwePacketLossEvent(int32_t bitrate,
uint8_t fraction_loss,
int32_t total_packets) override;
private:
void StoreEvent(std::unique_ptr<rtclog::Event>* event);
// Message queue for passing control messages to the logging thread.
SwapQueue<RtcEventLogHelperThread::ControlMessage> message_queue_;
// Message queue for passing events to the logging thread.
SwapQueue<std::unique_ptr<rtclog::Event> > event_queue_;
const Clock* const clock_;
RtcEventLogHelperThread helper_thread_;
rtc::ThreadChecker thread_checker_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtcEventLogImpl);
};
namespace {
// The functions in this namespace convert enums from the runtime format
// that the rest of the WebRtc project can use, to the corresponding
// serialized enum which is defined by the protobuf.
rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) {
switch (rtcp_mode) {
case RtcpMode::kCompound:
return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
case RtcpMode::kReducedSize:
return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE;
case RtcpMode::kOff:
RTC_NOTREACHED();
return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
}
RTC_NOTREACHED();
return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
}
rtclog::MediaType ConvertMediaType(MediaType media_type) {
switch (media_type) {
case MediaType::ANY:
return rtclog::MediaType::ANY;
case MediaType::AUDIO:
return rtclog::MediaType::AUDIO;
case MediaType::VIDEO:
return rtclog::MediaType::VIDEO;
case MediaType::DATA:
return rtclog::MediaType::DATA;
}
RTC_NOTREACHED();
return rtclog::ANY;
}
// The RTP and RTCP buffers reserve space for twice the expected number of
// sent packets because they also contain received packets.
static const int kEventsPerSecond = 1000;
static const int kControlMessagesPerSecond = 10;
} // namespace
// RtcEventLogImpl member functions.
RtcEventLogImpl::RtcEventLogImpl(const Clock* clock)
// Allocate buffers for roughly one second of history.
: message_queue_(kControlMessagesPerSecond),
event_queue_(kEventsPerSecond),
clock_(clock),
helper_thread_(&message_queue_,
&event_queue_,
clock),
thread_checker_() {
thread_checker_.DetachFromThread();
}
RtcEventLogImpl::~RtcEventLogImpl() {
// The RtcEventLogHelperThread destructor closes the file
// and waits for the thread to terminate.
}
bool RtcEventLogImpl::StartLogging(const std::string& file_name,
int64_t max_size_bytes) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RtcEventLogHelperThread::ControlMessage message;
message.message_type = RtcEventLogHelperThread::ControlMessage::START_FILE;
message.max_size_bytes = max_size_bytes <= 0
? std::numeric_limits<int64_t>::max()
: max_size_bytes;
message.start_time = clock_->TimeInMicroseconds();
message.stop_time = std::numeric_limits<int64_t>::max();
message.file.reset(FileWrapper::Create());
if (!message.file->OpenFile(file_name.c_str(), false)) {
LOG(LS_ERROR) << "Can't open file. WebRTC event log not started.";
return false;
}
if (!message_queue_.Insert(&message)) {
LOG(LS_ERROR) << "Message queue full. Can't start logging.";
return false;
}
helper_thread_.SignalNewEvent();
LOG(LS_INFO) << "Starting WebRTC event log.";
return true;
}
bool RtcEventLogImpl::StartLogging(rtc::PlatformFile platform_file,
int64_t max_size_bytes) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RtcEventLogHelperThread::ControlMessage message;
message.message_type = RtcEventLogHelperThread::ControlMessage::START_FILE;
message.max_size_bytes = max_size_bytes <= 0
? std::numeric_limits<int64_t>::max()
: max_size_bytes;
message.start_time = clock_->TimeInMicroseconds();
message.stop_time = std::numeric_limits<int64_t>::max();
message.file.reset(FileWrapper::Create());
FILE* file_handle = rtc::FdopenPlatformFileForWriting(platform_file);
if (!file_handle) {
LOG(LS_ERROR) << "Can't open file. WebRTC event log not started.";
// Even though we failed to open a FILE*, the platform_file is still open
// and needs to be closed.
if (!rtc::ClosePlatformFile(platform_file)) {
LOG(LS_ERROR) << "Can't close file.";
}
return false;
}
if (!message.file->OpenFromFileHandle(file_handle)) {
LOG(LS_ERROR) << "Can't open file. WebRTC event log not started.";
return false;
}
if (!message_queue_.Insert(&message)) {
LOG(LS_ERROR) << "Message queue full. Can't start logging.";
return false;
}
helper_thread_.SignalNewEvent();
LOG(LS_INFO) << "Starting WebRTC event log.";
return true;
}
void RtcEventLogImpl::StopLogging() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RtcEventLogHelperThread::ControlMessage message;
message.message_type = RtcEventLogHelperThread::ControlMessage::STOP_FILE;
message.stop_time = clock_->TimeInMicroseconds();
while (!message_queue_.Insert(&message)) {
// TODO(terelius): We would like to have a blocking Insert function in the
// SwapQueue, but for the time being we will just clear any previous
// messages.
// Since StopLogging waits for the thread, it is essential that we don't
// clear any STOP_FILE messages. To ensure that there is only one call at a
// time, we require that all calls to StopLogging are made on the same
// thread.
LOG(LS_ERROR) << "Message queue full. Clearing queue to stop logging.";
message_queue_.Clear();
}
LOG(LS_INFO) << "Stopping WebRTC event log.";
helper_thread_.WaitForFileFinished();
}
void RtcEventLogImpl::LogVideoReceiveStreamConfig(
const VideoReceiveStream::Config& config) {
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
event->set_timestamp_us(clock_->TimeInMicroseconds());
event->set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
rtclog::VideoReceiveConfig* receiver_config =
event->mutable_video_receiver_config();
receiver_config->set_remote_ssrc(config.rtp.remote_ssrc);
receiver_config->set_local_ssrc(config.rtp.local_ssrc);
receiver_config->set_rtcp_mode(ConvertRtcpMode(config.rtp.rtcp_mode));
receiver_config->set_remb(config.rtp.remb);
for (const auto& kv : config.rtp.rtx) {
rtclog::RtxMap* rtx = receiver_config->add_rtx_map();
rtx->set_payload_type(kv.first);
rtx->mutable_config()->set_rtx_ssrc(kv.second.ssrc);
rtx->mutable_config()->set_rtx_payload_type(kv.second.payload_type);
}
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);
}
for (const auto& d : config.decoders) {
rtclog::DecoderConfig* decoder = receiver_config->add_decoders();
decoder->set_name(d.payload_name);
decoder->set_payload_type(d.payload_type);
}
StoreEvent(&event);
}
void RtcEventLogImpl::LogVideoSendStreamConfig(
const VideoSendStream::Config& config) {
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
event->set_timestamp_us(clock_->TimeInMicroseconds());
event->set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
rtclog::VideoSendConfig* sender_config = event->mutable_video_sender_config();
for (const auto& ssrc : config.rtp.ssrcs) {
sender_config->add_ssrcs(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);
}
for (const auto& rtx_ssrc : config.rtp.rtx.ssrcs) {
sender_config->add_rtx_ssrcs(rtx_ssrc);
}
sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type);
rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
encoder->set_name(config.encoder_settings.payload_name);
encoder->set_payload_type(config.encoder_settings.payload_type);
StoreEvent(&event);
}
void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length) {
// Read header length (in bytes) from packet data.
if (packet_length < 12u) {
return; // Don't read outside the packet.
}
const bool x = (header[0] & 0x10) != 0;
const uint8_t cc = header[0] & 0x0f;
size_t header_length = 12u + cc * 4u;
if (x) {
if (packet_length < 12u + cc * 4u + 4u) {
return; // Don't read outside the packet.
}
size_t x_len = ByteReader<uint16_t>::ReadBigEndian(header + 14 + cc * 4);
header_length += (x_len + 1) * 4;
}
std::unique_ptr<rtclog::Event> rtp_event(new rtclog::Event());
rtp_event->set_timestamp_us(clock_->TimeInMicroseconds());
rtp_event->set_type(rtclog::Event::RTP_EVENT);
rtp_event->mutable_rtp_packet()->set_incoming(direction == kIncomingPacket);
rtp_event->mutable_rtp_packet()->set_type(ConvertMediaType(media_type));
rtp_event->mutable_rtp_packet()->set_packet_length(packet_length);
rtp_event->mutable_rtp_packet()->set_header(header, header_length);
StoreEvent(&rtp_event);
}
void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
MediaType media_type,
const uint8_t* packet,
size_t length) {
std::unique_ptr<rtclog::Event> rtcp_event(new rtclog::Event());
rtcp_event->set_timestamp_us(clock_->TimeInMicroseconds());
rtcp_event->set_type(rtclog::Event::RTCP_EVENT);
rtcp_event->mutable_rtcp_packet()->set_incoming(direction == kIncomingPacket);
rtcp_event->mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
RTCPUtility::RtcpCommonHeader header;
const uint8_t* block_begin = packet;
const uint8_t* packet_end = packet + length;
RTC_DCHECK(length <= IP_PACKET_SIZE);
uint8_t buffer[IP_PACKET_SIZE];
uint32_t buffer_length = 0;
while (block_begin < packet_end) {
if (!RtcpParseCommonHeader(block_begin, packet_end - block_begin,
&header)) {
break; // Incorrect message header.
}
uint32_t block_size = header.BlockSize();
switch (header.packet_type) {
case RTCPUtility::PT_SR:
FALLTHROUGH();
case RTCPUtility::PT_RR:
FALLTHROUGH();
case RTCPUtility::PT_BYE:
FALLTHROUGH();
case RTCPUtility::PT_IJ:
FALLTHROUGH();
case RTCPUtility::PT_RTPFB:
FALLTHROUGH();
case RTCPUtility::PT_PSFB:
FALLTHROUGH();
case RTCPUtility::PT_XR:
// We log sender reports, receiver reports, bye messages
// inter-arrival jitter, third-party loss reports, payload-specific
// feedback and extended reports.
memcpy(buffer + buffer_length, block_begin, block_size);
buffer_length += block_size;
break;
case RTCPUtility::PT_SDES:
FALLTHROUGH();
case RTCPUtility::PT_APP:
FALLTHROUGH();
default:
// We don't log sender descriptions, application defined messages
// or message blocks of unknown type.
break;
}
block_begin += block_size;
}
rtcp_event->mutable_rtcp_packet()->set_packet_data(buffer, buffer_length);
StoreEvent(&rtcp_event);
}
void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) {
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
event->set_timestamp_us(clock_->TimeInMicroseconds());
event->set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
auto playout_event = event->mutable_audio_playout_event();
playout_event->set_local_ssrc(ssrc);
StoreEvent(&event);
}
void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate,
uint8_t fraction_loss,
int32_t total_packets) {
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
event->set_timestamp_us(clock_->TimeInMicroseconds());
event->set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT);
auto bwe_event = event->mutable_bwe_packet_loss_event();
bwe_event->set_bitrate(bitrate);
bwe_event->set_fraction_loss(fraction_loss);
bwe_event->set_total_packets(total_packets);
StoreEvent(&event);
}
void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event>* event) {
if (!event_queue_.Insert(event)) {
LOG(LS_ERROR) << "WebRTC event log queue full. Dropping event.";
}
helper_thread_.SignalNewEvent();
}
bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
rtclog::EventStream* result) {
char tmp_buffer[1024];
int bytes_read = 0;
std::unique_ptr<FileWrapper> dump_file(FileWrapper::Create());
if (!dump_file->OpenFile(file_name.c_str(), true)) {
return false;
}
std::string dump_buffer;
while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) {
dump_buffer.append(tmp_buffer, bytes_read);
}
dump_file->CloseFile();
return result->ParseFromString(dump_buffer);
}
#endif // ENABLE_RTC_EVENT_LOG
// RtcEventLog member functions.
std::unique_ptr<RtcEventLog> RtcEventLog::Create(const Clock* clock) {
#ifdef ENABLE_RTC_EVENT_LOG
return std::unique_ptr<RtcEventLog>(new RtcEventLogImpl(clock));
#else
return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
#endif // ENABLE_RTC_EVENT_LOG
}
} // namespace webrtc
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