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This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.

Browse Source 
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader.
Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead.

BUG=webrtc:7538
TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc

Review-Url: https://codereview.webrtc.org/2855143002
Cr-Commit-Position: refs/heads/master@{#18324}
This commit is contained in:
perkj
2017-05-30 03:52:10 -07:00
committed by Commit Bot
parent 367aba92bf
commit 77cd58e140
21 changed files with 295 additions and 327 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/call/call.cc
View File

@ -1214,7 +1214,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
}
if (rtcp_delivered)
event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet, length);
event_log_->LogRtcpPacket(kIncomingPacket, packet, length);
return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
}
@ -1242,14 +1242,14 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
if (audio_rtp_demuxer_.OnRtpPacket(*parsed_packet)) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
event_log_->LogRtpHeader(kIncomingPacket, packet, length);
return DELIVERY_OK;
}
} else if (media_type == MediaType::VIDEO) {
if (video_rtp_demuxer_.OnRtpPacket(*parsed_packet)) {
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
event_log_->LogRtpHeader(kIncomingPacket, packet, length);
return DELIVERY_OK;
}
}

6
webrtc/logging/BUILD.gn
View File

@ -28,8 +28,8 @@ rtc_source_set("rtc_event_log_api") {
]
deps = [
"..:video_stream_api",
"..:webrtc_common",
"../base:rtc_base_approved",
"../call:call_interfaces",
]
}
@ -48,7 +48,6 @@ rtc_static_library("rtc_event_log_impl") {
"..:webrtc_common",
"../base:protobuf_utils",
"../base:rtc_base_approved",
"../call:call_interfaces",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp",
@ -83,7 +82,6 @@ if (rtc_enable_protobuf) {
":rtc_event_log_api",
":rtc_event_log_proto",
"..:webrtc_common",
"../call:call_interfaces",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp",
@ -138,7 +136,6 @@ if (rtc_enable_protobuf) {
":rtc_event_log_impl",
":rtc_event_log_parser",
"../base:rtc_base_approved",
"../call:call_interfaces",
"../modules/rtp_rtcp:rtp_rtcp",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
@ -162,7 +159,6 @@ if (rtc_enable_protobuf) {
":rtc_event_log_impl",
":rtc_event_log_parser",
"../base:rtc_base_approved",
"../call:call_interfaces",
# TODO(kwiberg): Remove this dependency.
"../api/audio_codecs:audio_codecs_api",

9
webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
View File

@ -41,22 +41,19 @@ class MockRtcEventLog : public RtcEventLog {
MOCK_METHOD1(LogAudioSendStreamConfig,
void(const rtclog::StreamConfig& config));
MOCK_METHOD4(LogRtpHeader,
MOCK_METHOD3(LogRtpHeader,
void(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length));
MOCK_METHOD5(LogRtpHeader,
MOCK_METHOD4(LogRtpHeader,
void(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id));
MOCK_METHOD4(LogRtcpPacket,
MOCK_METHOD3(LogRtcpPacket,
void(PacketDirection direction,
MediaType media_type,
const uint8_t* packet,
size_t length));

27
webrtc/logging/rtc_event_log/rtc_event_log.cc
View File

@ -21,7 +21,6 @@
#include "webrtc/base/swap_queue.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
@ -67,16 +66,13 @@ class RtcEventLogImpl final : public RtcEventLog {
void LogAudioReceiveStreamConfig(const rtclog::StreamConfig& config) override;
void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override;
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length) override;
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) override;
void LogRtcpPacket(PacketDirection direction,
MediaType media_type,
const uint8_t* packet,
size_t length) override;
void LogAudioPlayout(uint32_t ssrc) override;
@ -132,21 +128,6 @@ rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) {
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;
}
rtclog::DelayBasedBweUpdate::DetectorState ConvertDetectorState(
BandwidthUsage state) {
switch (state) {
@ -390,15 +371,12 @@ void RtcEventLogImpl::LogAudioSendStreamConfig(
}
void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length) {
LogRtpHeader(direction, media_type, header, packet_length,
PacedPacketInfo::kNotAProbe);
LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe);
}
void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) {
@ -422,7 +400,6 @@ void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
rtp_event->set_timestamp_us(rtc::TimeMicros());
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);
if (probe_cluster_id != PacedPacketInfo::kNotAProbe)
@ -431,14 +408,12 @@ void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
}
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(rtc::TimeMicros());
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));
rtcp::CommonHeader header;
const uint8_t* block_begin = packet;

11
webrtc/logging/rtc_event_log/rtc_event_log.h
View File

@ -16,10 +16,7 @@
#include <vector>
#include "webrtc/base/platform_file.h"
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/audio_send_stream.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
#include "webrtc/config.h"
namespace webrtc {
@ -129,21 +126,18 @@ class RtcEventLog {
// Logs the header of an incoming or outgoing RTP packet. packet_length
// is the total length of the packet, including both header and payload.
virtual void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length) = 0;
// Same as above but used on the sender side to log packets that are part of
// a probe cluster.
virtual void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) = 0;
// Logs an incoming or outgoing RTCP packet.
virtual void LogRtcpPacket(PacketDirection direction,
MediaType media_type,
const uint8_t* packet,
size_t length) = 0;
@ -204,16 +198,13 @@ class RtcEventLogNullImpl : public RtcEventLog {
const rtclog::StreamConfig& config) override {}
void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {}
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length) override {}
void LogRtpHeader(PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) override {}
void LogRtcpPacket(PacketDirection direction,
MediaType media_type,
const uint8_t* packet,
size_t length) override {}
void LogAudioPlayout(uint32_t ssrc) override {}

6
webrtc/logging/rtc_event_log/rtc_event_log.proto
View File

@ -86,8 +86,7 @@ message RtpPacket {
// required - True if the packet is incoming w.r.t. the user logging the data
optional bool incoming = 1;
// required
optional MediaType type = 2;
optional MediaType type = 2 [deprecated = true];
// required - The size of the packet including both payload and header.
optional uint32 packet_length = 3;
@ -105,8 +104,7 @@ message RtcpPacket {
// required - True if the packet is incoming w.r.t. the user logging the data
optional bool incoming = 1;
// required
optional MediaType type = 2;
optional MediaType type = 2 [deprecated = true];
// required - The whole packet including both payload and header.
optional bytes packet_data = 3;

43
webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
View File

@ -15,14 +15,16 @@
#include "gflags/gflags.h"
#include "webrtc/base/checks.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/test/rtp_file_writer.h"
namespace {
using MediaType = webrtc::ParsedRtcEventLog::MediaType;
DEFINE_bool(noaudio,
false,
"Excludes audio packets from the converted RTPdump file.");
@ -118,21 +120,28 @@ int main(int argc, char* argv[]) {
parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
webrtc::test::RtpPacket packet;
webrtc::PacketDirection direction;
webrtc::MediaType media_type;
parsed_stream.GetRtpHeader(i, &direction, &media_type, packet.data,
&packet.length, &packet.original_length);
parsed_stream.GetRtpHeader(i, &direction, packet.data, &packet.length,
&packet.original_length);
if (packet.original_length > packet.length)
header_only = true;
packet.time_ms = parsed_stream.GetTimestamp(i) / 1000;
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet.data,
packet.length);
// TODO(terelius): Maybe add a flag to dump outgoing traffic instead?
if (direction == webrtc::kOutgoingPacket)
continue;
if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
webrtc::RTPHeader parsed_header;
rtp_parser.Parse(&parsed_header);
MediaType media_type =
parsed_stream.GetMediaType(parsed_header.ssrc, direction);
if (FLAGS_noaudio && media_type == MediaType::AUDIO)
continue;
if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
if (FLAGS_novideo && media_type == MediaType::VIDEO)
continue;
if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
if (FLAGS_nodata && media_type == MediaType::DATA)
continue;
if (!FLAGS_ssrc.empty()) {
const uint32_t packet_ssrc =
@ -150,9 +159,7 @@ int main(int argc, char* argv[]) {
webrtc::ParsedRtcEventLog::RTCP_EVENT) {
webrtc::test::RtpPacket packet;
webrtc::PacketDirection direction;
webrtc::MediaType media_type;
parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet.data,
&packet.length);
parsed_stream.GetRtcpPacket(i, &direction, packet.data, &packet.length);
// For RTCP packets the original_length should be set to 0 in the
// RTPdump format.
packet.original_length = 0;
@ -161,16 +168,20 @@ int main(int argc, char* argv[]) {
// TODO(terelius): Maybe add a flag to dump outgoing traffic instead?
if (direction == webrtc::kOutgoingPacket)
continue;
if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
// Note that |packet_ssrc| is the sender SSRC. An RTCP message may contain
// report blocks for many streams, thus several SSRCs and they doen't
// necessarily have to be of the same media type.
const uint32_t packet_ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(
reinterpret_cast<const uint8_t*>(packet.data + 4));
MediaType media_type = parsed_stream.GetMediaType(packet_ssrc, direction);
if (FLAGS_noaudio && media_type == MediaType::AUDIO)
continue;
if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
if (FLAGS_novideo && media_type == MediaType::VIDEO)
continue;
if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
if (FLAGS_nodata && media_type == MediaType::DATA)
continue;
if (!FLAGS_ssrc.empty()) {
const uint32_t packet_ssrc =
webrtc::ByteReader<uint32_t>::ReadBigEndian(
reinterpret_cast<const uint8_t*>(packet.data + 4));
if (packet_ssrc != ssrc_filter)
continue;
}

203
webrtc/logging/rtc_event_log/rtc_event_log2text.cc
View File

@ -14,7 +14,6 @@
#include "gflags/gflags.h"
#include "webrtc/base/checks.h"
#include "webrtc/call/call.h"
#include "webrtc/common_types.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
@ -52,6 +51,8 @@ DEFINE_string(ssrc,
"Print only packets with this SSRC (decimal or hex, the latter "
"starting with 0x).");
using MediaType = webrtc::ParsedRtcEventLog::MediaType;
static uint32_t filtered_ssrc = 0;
// Parses the input string for a valid SSRC. If a valid SSRC is found, it is
@ -73,44 +74,18 @@ bool ParseSsrc(std::string str) {
return str.empty() || (!ss.fail() && ss.eof());
}
// Struct used for storing SSRCs used in a Stream.
struct Stream {
Stream(uint32_t ssrc,
webrtc::MediaType media_type,
webrtc::PacketDirection direction)
: ssrc(ssrc), media_type(media_type), direction(direction) {}
uint32_t ssrc;
webrtc::MediaType media_type;
webrtc::PacketDirection direction;
};
// All configured streams found in the event log.
std::vector<Stream> global_streams;
// Returns the MediaType for registered SSRCs. Search from the end to use last
// registered types first.
webrtc::MediaType GetMediaType(uint32_t ssrc,
webrtc::PacketDirection direction) {
for (auto rit = global_streams.rbegin(); rit != global_streams.rend();
++rit) {
if (rit->ssrc == ssrc && rit->direction == direction)
return rit->media_type;
}
return webrtc::MediaType::ANY;
}
bool ExcludePacket(webrtc::PacketDirection direction,
webrtc::MediaType media_type,
MediaType media_type,
uint32_t packet_ssrc) {
if (FLAGS_nooutgoing && direction == webrtc::kOutgoingPacket)
return true;
if (FLAGS_noincoming && direction == webrtc::kIncomingPacket)
return true;
if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
if (FLAGS_noaudio && media_type == MediaType::AUDIO)
return true;
if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
if (FLAGS_novideo && media_type == MediaType::VIDEO)
return true;
if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
if (FLAGS_nodata && media_type == MediaType::DATA)
return true;
if (!FLAGS_ssrc.empty() && packet_ssrc != filtered_ssrc)
return true;
@ -118,23 +93,23 @@ bool ExcludePacket(webrtc::PacketDirection direction,
}
const char* StreamInfo(webrtc::PacketDirection direction,
webrtc::MediaType media_type) {
MediaType media_type) {
if (direction == webrtc::kOutgoingPacket) {
if (media_type == webrtc::MediaType::AUDIO)
if (media_type == MediaType::AUDIO)
return "(out,audio)";
else if (media_type == webrtc::MediaType::VIDEO)
else if (media_type == MediaType::VIDEO)
return "(out,video)";
else if (media_type == webrtc::MediaType::DATA)
else if (media_type == MediaType::DATA)
return "(out,data)";
else
return "(out)";
}
if (direction == webrtc::kIncomingPacket) {
if (media_type == webrtc::MediaType::AUDIO)
if (media_type == MediaType::AUDIO)
return "(in,audio)";
else if (media_type == webrtc::MediaType::VIDEO)
else if (media_type == MediaType::VIDEO)
return "(in,video)";
else if (media_type == webrtc::MediaType::DATA)
else if (media_type == MediaType::DATA)
return "(in,data)";
else
return "(in)";
@ -142,13 +117,15 @@ const char* StreamInfo(webrtc::PacketDirection direction,
return "(unknown)";
}
void PrintSenderReport(const webrtc::rtcp::CommonHeader& rtcp_block,
void PrintSenderReport(const webrtc::ParsedRtcEventLog& parsed_stream,
const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
webrtc::PacketDirection direction) {
webrtc::rtcp::SenderReport sr;
if (!sr.Parse(rtcp_block))
return;
webrtc::MediaType media_type = GetMediaType(sr.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(sr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, sr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -157,13 +134,15 @@ void PrintSenderReport(const webrtc::rtcp::CommonHeader& rtcp_block,
<< "\ttimestamp=" << sr.rtp_timestamp() << std::endl;
}
void PrintReceiverReport(const webrtc::rtcp::CommonHeader& rtcp_block,
void PrintReceiverReport(const webrtc::ParsedRtcEventLog& parsed_stream,
const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
webrtc::PacketDirection direction) {
webrtc::rtcp::ReceiverReport rr;
if (!rr.Parse(rtcp_block))
return;
webrtc::MediaType media_type = GetMediaType(rr.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(rr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, rr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -171,13 +150,15 @@ void PrintReceiverReport(const webrtc::rtcp::CommonHeader& rtcp_block,
<< "\tssrc=" << rr.sender_ssrc() << std::endl;
}
void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
void PrintXr(const webrtc::ParsedRtcEventLog& parsed_stream,
const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
webrtc::PacketDirection direction) {
webrtc::rtcp::ExtendedReports xr;
if (!xr.Parse(rtcp_block))
return;
webrtc::MediaType media_type = GetMediaType(xr.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(xr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, xr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -189,18 +170,20 @@ void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
webrtc::PacketDirection direction) {
std::cout << log_timestamp << "\t"
<< "RTCP_SDES" << StreamInfo(direction, webrtc::MediaType::ANY)
<< "RTCP_SDES" << StreamInfo(direction, MediaType::ANY)
<< std::endl;
RTC_NOTREACHED() << "SDES should have been redacted when writing the log";
}
void PrintBye(const webrtc::rtcp::CommonHeader& rtcp_block,
void PrintBye(const webrtc::ParsedRtcEventLog& parsed_stream,
const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
webrtc::PacketDirection direction) {
webrtc::rtcp::Bye bye;
if (!bye.Parse(rtcp_block))
return;
webrtc::MediaType media_type = GetMediaType(bye.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(bye.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, bye.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -208,7 +191,8 @@ void PrintBye(const webrtc::rtcp::CommonHeader& rtcp_block,
<< "\tssrc=" << bye.sender_ssrc() << std::endl;
}
void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
void PrintRtpFeedback(const webrtc::ParsedRtcEventLog& parsed_stream,
const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
webrtc::PacketDirection direction) {
switch (rtcp_block.fmt()) {
@ -216,8 +200,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Nack nack;
if (!nack.Parse(rtcp_block))
return;
webrtc::MediaType media_type =
GetMediaType(nack.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(nack.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, nack.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -229,8 +213,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Tmmbr tmmbr;
if (!tmmbr.Parse(rtcp_block))
return;
webrtc::MediaType media_type =
GetMediaType(tmmbr.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(tmmbr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, tmmbr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -242,8 +226,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Tmmbn tmmbn;
if (!tmmbn.Parse(rtcp_block))
return;
webrtc::MediaType media_type =
GetMediaType(tmmbn.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(tmmbn.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, tmmbn.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -255,8 +239,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::RapidResyncRequest sr_req;
if (!sr_req.Parse(rtcp_block))
return;
webrtc::MediaType media_type =
GetMediaType(sr_req.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(sr_req.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, sr_req.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -268,8 +252,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::TransportFeedback transport_feedback;
if (!transport_feedback.Parse(rtcp_block))
return;
webrtc::MediaType media_type =
GetMediaType(transport_feedback.sender_ssrc(), direction);
MediaType media_type = parsed_stream.GetMediaType(
transport_feedback.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type,
transport_feedback.sender_ssrc()))
return;
@ -283,7 +267,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
}
}
void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
void PrintPsFeedback(const webrtc::ParsedRtcEventLog& parsed_stream,
const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
webrtc::PacketDirection direction) {
switch (rtcp_block.fmt()) {
@ -291,7 +276,8 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Pli pli;
if (!pli.Parse(rtcp_block))
return;
webrtc::MediaType media_type = GetMediaType(pli.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(pli.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, pli.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -303,7 +289,8 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Fir fir;
if (!fir.Parse(rtcp_block))
return;
webrtc::MediaType media_type = GetMediaType(fir.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(fir.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, fir.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -315,8 +302,8 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
webrtc::rtcp::Remb remb;
if (!remb.Parse(rtcp_block))
return;
webrtc::MediaType media_type =
GetMediaType(remb.sender_ssrc(), direction);
MediaType media_type =
parsed_stream.GetMediaType(remb.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, remb.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@ -362,67 +349,41 @@ int main(int argc, char* argv[]) {
}
for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
if (parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config;
parsed_stream.GetVideoReceiveConfig(i, &config);
global_streams.emplace_back(config.remote_ssrc,
webrtc::MediaType::VIDEO,
webrtc::kIncomingPacket);
global_streams.emplace_back(config.local_ssrc,
webrtc::MediaType::VIDEO,
webrtc::kOutgoingPacket);
if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming) {
std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
<< "\tssrc=" << config.remote_ssrc
<< "\tfeedback_ssrc=" << config.local_ssrc << std::endl;
}
}
if (parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config;
parsed_stream.GetVideoSendConfig(i, &config);
global_streams.emplace_back(config.local_ssrc, webrtc::MediaType::VIDEO,
webrtc::kOutgoingPacket);
global_streams.emplace_back(config.rtx_ssrc, webrtc::MediaType::VIDEO,
webrtc::kOutgoingPacket);
if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing) {
std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
std::cout << "\tssrcs=" << config.local_ssrc;
std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
std::cout << std::endl;
}
}
if (parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config;
parsed_stream.GetAudioReceiveConfig(i, &config);
global_streams.emplace_back(config.remote_ssrc,
webrtc::MediaType::AUDIO,
webrtc::kIncomingPacket);
global_streams.emplace_back(config.local_ssrc,
webrtc::MediaType::AUDIO,
webrtc::kOutgoingPacket);
if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming) {
std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
<< "\tssrc=" << config.remote_ssrc
<< "\tfeedback_ssrc=" << config.local_ssrc << std::endl;
}
std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
<< "\tssrc=" << config.remote_ssrc
<< "\tfeedback_ssrc=" << config.local_ssrc << std::endl;
}
if (parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing &&
parsed_stream.GetEventType(i) ==
webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
webrtc::rtclog::StreamConfig config;
parsed_stream.GetAudioSendConfig(i, &config);
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.local_ssrc << std::endl;
}
std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
<< "\tssrc=" << config.local_ssrc << std::endl;
}
if (!FLAGS_nortp &&
parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
@ -430,15 +391,16 @@ int main(int argc, char* argv[]) {
size_t total_length;
uint8_t header[IP_PACKET_SIZE];
webrtc::PacketDirection direction;
webrtc::MediaType media_type;
parsed_stream.GetRtpHeader(i, &direction, &media_type, header,
&header_length, &total_length);
parsed_stream.GetRtpHeader(i, &direction, header, &header_length,
&total_length);
// Parse header to get SSRC and RTP time.
webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
webrtc::RTPHeader parsed_header;
rtp_parser.Parse(&parsed_header);
media_type = GetMediaType(parsed_header.ssrc, direction);
MediaType media_type =
parsed_stream.GetMediaType(parsed_header.ssrc, direction);
if (ExcludePacket(direction, media_type, parsed_header.ssrc))
continue;
@ -454,8 +416,7 @@ int main(int argc, char* argv[]) {
size_t length;
uint8_t packet[IP_PACKET_SIZE];
webrtc::PacketDirection direction;
webrtc::MediaType media_type;
parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet, &length);
parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
webrtc::rtcp::CommonHeader rtcp_block;
const uint8_t* packet_end = packet + length;
@ -470,25 +431,29 @@ int main(int argc, char* argv[]) {
uint64_t log_timestamp = parsed_stream.GetTimestamp(i);
switch (rtcp_block.type()) {
case webrtc::rtcp::SenderReport::kPacketType:
PrintSenderReport(rtcp_block, log_timestamp, direction);
PrintSenderReport(parsed_stream, rtcp_block, log_timestamp,
direction);
break;
case webrtc::rtcp::ReceiverReport::kPacketType:
PrintReceiverReport(rtcp_block, log_timestamp, direction);
PrintReceiverReport(parsed_stream, rtcp_block, log_timestamp,
direction);
break;
case webrtc::rtcp::Sdes::kPacketType:
PrintSdes(rtcp_block, log_timestamp, direction);
break;
case webrtc::rtcp::ExtendedReports::kPacketType:
PrintXr(rtcp_block, log_timestamp, direction);
PrintXr(parsed_stream, rtcp_block, log_timestamp, direction);
break;
case webrtc::rtcp::Bye::kPacketType:
PrintBye(rtcp_block, log_timestamp, direction);
PrintBye(parsed_stream, rtcp_block, log_timestamp, direction);
break;
case webrtc::rtcp::Rtpfb::kPacketType:
PrintRtpFeedback(rtcp_block, log_timestamp, direction);
PrintRtpFeedback(parsed_stream, rtcp_block, log_timestamp,
direction);
break;
case webrtc::rtcp::Psfb::kPacketType:
PrintPsFeedback(rtcp_block, log_timestamp, direction);
PrintPsFeedback(parsed_stream, rtcp_block, log_timestamp,
direction);
break;
default:
break;

110
webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
View File

@ -22,7 +22,6 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/protobuf_utils.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
@ -31,21 +30,6 @@
namespace webrtc {
namespace {
MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
switch (media_type) {
case rtclog::MediaType::ANY:
return MediaType::ANY;
case rtclog::MediaType::AUDIO:
return MediaType::AUDIO;
case rtclog::MediaType::VIDEO:
return MediaType::VIDEO;
case rtclog::MediaType::DATA:
return MediaType::DATA;
}
RTC_NOTREACHED();
return MediaType::ANY;
}
RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
switch (rtcp_mode) {
case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
@ -179,7 +163,8 @@ bool ParsedRtcEventLog::ParseStream(std::istream& stream) {
// Read the next message tag. The tag number is defined as
// (fieldnumber << 3) | wire_type. In our case, the field number is
// supposed to be 1 and the wire type for an length-delimited field is 2.
// supposed to be 1 and the wire type for an
// length-delimited field is 2.
const uint64_t kExpectedTag = (1 << 3) | 2;
std::tie(tag, success) = ParseVarInt(stream);
if (!success) {
@ -213,6 +198,48 @@ bool ParsedRtcEventLog::ParseStream(std::istream& stream) {
LOG(LS_WARNING) << "Failed to parse protobuf message.";
return false;
}
EventType type = GetRuntimeEventType(event.type());
switch (type) {
case VIDEO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config;
GetVideoReceiveConfig(event, &config);
streams_.emplace_back(config.remote_ssrc, MediaType::VIDEO,
kIncomingPacket);
streams_.emplace_back(config.local_ssrc, MediaType::VIDEO,
kOutgoingPacket);
break;
}
case VIDEO_SENDER_CONFIG_EVENT: {
rtclog::StreamConfig config;
GetVideoSendConfig(event, &config);
streams_.emplace_back(config.local_ssrc, MediaType::VIDEO,
kOutgoingPacket);
streams_.emplace_back(config.rtx_ssrc, MediaType::VIDEO,
kOutgoingPacket);
break;
}
case AUDIO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config;
GetAudioReceiveConfig(event, &config);
streams_.emplace_back(config.remote_ssrc, MediaType::AUDIO,
kIncomingPacket);
streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
kOutgoingPacket);
break;
}
case AUDIO_SENDER_CONFIG_EVENT: {
rtclog::StreamConfig config;
GetAudioSendConfig(event, &config);
streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
kOutgoingPacket);
break;
}
default:
break;
}
events_.push_back(event);
}
}
@ -239,7 +266,6 @@ ParsedRtcEventLog::EventType ParsedRtcEventLog::GetEventType(
// The header must have space for at least IP_PACKET_SIZE bytes.
void ParsedRtcEventLog::GetRtpHeader(size_t index,
PacketDirection* incoming,
MediaType* media_type,
uint8_t* header,
size_t* header_length,
size_t* total_length) const {
@ -254,11 +280,6 @@ void ParsedRtcEventLog::GetRtpHeader(size_t index,
if (incoming != nullptr) {
*incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
}
// Get media type.
RTC_CHECK(rtp_packet.has_type());
if (media_type != nullptr) {
*media_type = GetRuntimeMediaType(rtp_packet.type());
}
// Get packet length.
RTC_CHECK(rtp_packet.has_packet_length());
if (total_length != nullptr) {
@ -282,7 +303,6 @@ void ParsedRtcEventLog::GetRtpHeader(size_t index,
// The packet must have space for at least IP_PACKET_SIZE bytes.
void ParsedRtcEventLog::GetRtcpPacket(size_t index,
PacketDirection* incoming,
MediaType* media_type,
uint8_t* packet,
size_t* length) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
@ -296,11 +316,6 @@ void ParsedRtcEventLog::GetRtcpPacket(size_t index,
if (incoming != nullptr) {
*incoming = rtcp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
}
// Get media type.
RTC_CHECK(rtcp_packet.has_type());
if (media_type != nullptr) {
*media_type = GetRuntimeMediaType(rtcp_packet.type());
}
// Get packet length.
RTC_CHECK(rtcp_packet.has_packet_data());
if (length != nullptr) {
@ -319,7 +334,12 @@ void ParsedRtcEventLog::GetVideoReceiveConfig(
size_t index,
rtclog::StreamConfig* config) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
GetVideoReceiveConfig(events_[index], config);
}
void ParsedRtcEventLog::GetVideoReceiveConfig(
const rtclog::Event& event,
rtclog::StreamConfig* config) const {
RTC_CHECK(config != nullptr);
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
@ -381,7 +401,10 @@ void ParsedRtcEventLog::GetVideoReceiveConfig(
void ParsedRtcEventLog::GetVideoSendConfig(size_t index,
rtclog::StreamConfig* config) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
GetVideoSendConfig(events_[index], config);
}
void ParsedRtcEventLog::GetVideoSendConfig(const rtclog::Event& event,
rtclog::StreamConfig* config) const {
RTC_CHECK(config != nullptr);
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
@ -419,7 +442,12 @@ void ParsedRtcEventLog::GetAudioReceiveConfig(
size_t index,
rtclog::StreamConfig* config) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
GetAudioReceiveConfig(events_[index], config);
}
void ParsedRtcEventLog::GetAudioReceiveConfig(
const rtclog::Event& event,
rtclog::StreamConfig* config) const {
RTC_CHECK(config != nullptr);
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
@ -439,7 +467,11 @@ void ParsedRtcEventLog::GetAudioReceiveConfig(
void ParsedRtcEventLog::GetAudioSendConfig(size_t index,
rtclog::StreamConfig* config) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
GetAudioSendConfig(events_[index], config);
}
void ParsedRtcEventLog::GetAudioSendConfig(const rtclog::Event& event,
rtclog::StreamConfig* config) const {
RTC_CHECK(config != nullptr);
RTC_CHECK(event.has_type());
RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
@ -588,4 +620,16 @@ ParsedRtcEventLog::BweProbeResultEvent ParsedRtcEventLog::GetBweProbeResult(
return res;
}
// Returns the MediaType for registered SSRCs. Search from the end to use last
// registered types first.
ParsedRtcEventLog::MediaType ParsedRtcEventLog::GetMediaType(
uint32_t ssrc,
PacketDirection direction) const {
for (auto rit = streams_.rbegin(); rit != streams_.rend(); ++rit) {
if (rit->ssrc == ssrc && rit->direction == direction)
return rit->media_type;
}
return MediaType::ANY;
}
} // namespace webrtc

45
webrtc/logging/rtc_event_log/rtc_event_log_parser.h
View File

@ -74,6 +74,8 @@ class ParsedRtcEventLog {
BWE_PROBE_RESULT_EVENT = 18
};
enum class MediaType { ANY, AUDIO, VIDEO, DATA };
// Reads an RtcEventLog file and returns true if parsing was successful.
bool ParseFile(const std::string& file_name);
@ -92,25 +94,23 @@ class ParsedRtcEventLog {
// Reads the event type of the rtclog::Event at |index|.
EventType GetEventType(size_t index) const;
// Reads the header, direction, media type, header length and packet length
// from the RTP event at |index|, and stores the values in the corresponding
// output parameters. Each output parameter can be set to nullptr if that
// value isn't needed.
// Reads the header, direction, header length and packet length from the RTP
// event at |index|, and stores the values in the corresponding output
// parameters. Each output parameter can be set to nullptr if that value
// isn't needed.
// NB: The header must have space for at least IP_PACKET_SIZE bytes.
void GetRtpHeader(size_t index,
PacketDirection* incoming,
MediaType* media_type,
uint8_t* header,
size_t* header_length,
size_t* total_length) const;
// Reads packet, direction, media type and packet length from the RTCP event
// at |index|, and stores the values in the corresponding output parameters.
// Reads packet, direction and packet length from the RTCP event at |index|,
// and stores the values in the corresponding output parameters.
// Each output parameter can be set to nullptr if that value isn't needed.
// NB: The packet must have space for at least IP_PACKET_SIZE bytes.
void GetRtcpPacket(size_t index,
PacketDirection* incoming,
MediaType* media_type,
uint8_t* packet,
size_t* length) const;
@ -158,13 +158,36 @@ class ParsedRtcEventLog {
void GetAudioNetworkAdaptation(size_t index,
AudioEncoderRuntimeConfig* config) const;
ParsedRtcEventLog::BweProbeClusterCreatedEvent GetBweProbeClusterCreated(
size_t index) const;
BweProbeClusterCreatedEvent GetBweProbeClusterCreated(size_t index) const;
ParsedRtcEventLog::BweProbeResultEvent GetBweProbeResult(size_t index) const;
BweProbeResultEvent GetBweProbeResult(size_t index) const;
MediaType GetMediaType(uint32_t ssrc, PacketDirection direction) const;
private:
void GetVideoReceiveConfig(const rtclog::Event& event,
rtclog::StreamConfig* config) const;
void GetVideoSendConfig(const rtclog::Event& event,
rtclog::StreamConfig* config) const;
void GetAudioReceiveConfig(const rtclog::Event& event,
rtclog::StreamConfig* config) const;
void GetAudioSendConfig(const rtclog::Event& event,
rtclog::StreamConfig* config) const;
std::vector<rtclog::Event> events_;
struct Stream {
Stream(uint32_t ssrc,
MediaType media_type,
webrtc::PacketDirection direction)
: ssrc(ssrc), media_type(media_type), direction(direction) {}
uint32_t ssrc;
MediaType media_type;
webrtc::PacketDirection direction;
};
// All configured streams found in the event log.
std::vector<Stream> streams_;
};
} // namespace webrtc

17
webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
View File

@ -306,13 +306,11 @@ void LogSessionAndReadBack(size_t rtp_count,
for (size_t i = 1; i <= rtp_count; i++) {
log_dumper->LogRtpHeader(
(i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
(i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
rtp_packets[i - 1].data(), rtp_packets[i - 1].size());
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
if (i * rtcp_count >= rtcp_index * rtp_count) {
log_dumper->LogRtcpPacket(
(rtcp_index % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
rtcp_packets[rtcp_index - 1].data(),
rtcp_packets[rtcp_index - 1].size());
rtcp_index++;
@ -368,7 +366,6 @@ void LogSessionAndReadBack(size_t rtp_count,
RtcEventLogTestHelper::VerifyRtpEvent(
parsed_log, event_index,
(i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
(i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
rtp_packets[i - 1].data(), rtp_packets[i - 1].headers_size(),
rtp_packets[i - 1].size());
event_index++;
@ -376,7 +373,6 @@ void LogSessionAndReadBack(size_t rtp_count,
RtcEventLogTestHelper::VerifyRtcpEvent(
parsed_log, event_index,
rtcp_index % 2 == 0 ? kIncomingPacket : kOutgoingPacket,
rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
rtcp_packets[rtcp_index - 1].data(),
rtcp_packets[rtcp_index - 1].size());
event_index++;
@ -454,15 +450,15 @@ TEST(RtcEventLogTest, LogEventAndReadBack) {
fake_clock.SetTimeMicros(prng.Rand<uint32_t>());
std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
log_dumper->LogRtpHeader(kIncomingPacket, MediaType::VIDEO, rtp_packet.data(),
log_dumper->LogRtpHeader(kIncomingPacket, rtp_packet.data(),
rtp_packet.size());
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->StartLogging(temp_filename, 10000000);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->LogRtcpPacket(kOutgoingPacket, MediaType::VIDEO,
rtcp_packet.data(), rtcp_packet.size());
log_dumper->LogRtcpPacket(kOutgoingPacket, rtcp_packet.data(),
rtcp_packet.size());
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->StopLogging();
@ -478,12 +474,11 @@ TEST(RtcEventLogTest, LogEventAndReadBack) {
RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
RtcEventLogTestHelper::VerifyRtpEvent(
parsed_log, 1, kIncomingPacket, MediaType::VIDEO, rtp_packet.data(),
parsed_log, 1, kIncomingPacket, rtp_packet.data(),
rtp_packet.headers_size(), rtp_packet.size());
RtcEventLogTestHelper::VerifyRtcpEvent(parsed_log, 2, kOutgoingPacket,
MediaType::VIDEO, rtcp_packet.data(),
rtcp_packet.size());
RtcEventLogTestHelper::VerifyRtcpEvent(
parsed_log, 2, kOutgoingPacket, rtcp_packet.data(), rtcp_packet.size());
RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log, 3);

33
webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
View File

@ -30,20 +30,6 @@
namespace webrtc {
namespace {
MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
switch (media_type) {
case rtclog::MediaType::ANY:
return MediaType::ANY;
case rtclog::MediaType::AUDIO:
return MediaType::AUDIO;
case rtclog::MediaType::VIDEO:
return MediaType::VIDEO;
case rtclog::MediaType::DATA:
return MediaType::DATA;
}
RTC_NOTREACHED();
return MediaType::ANY;
}
BandwidthUsage GetRuntimeDetectorState(
rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
@ -367,7 +353,6 @@ void RtcEventLogTestHelper::VerifyAudioSendStreamConfig(
void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t header_size,
size_t total_size) {
@ -377,8 +362,6 @@ void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
ASSERT_TRUE(rtp_packet.has_incoming());
EXPECT_EQ(direction == kIncomingPacket, rtp_packet.incoming());
ASSERT_TRUE(rtp_packet.has_type());
EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type()));
ASSERT_TRUE(rtp_packet.has_packet_length());
EXPECT_EQ(total_size, rtp_packet.packet_length());
ASSERT_TRUE(rtp_packet.has_header());
@ -389,14 +372,11 @@ void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
// Check consistency of the parser.
PacketDirection parsed_direction;
MediaType parsed_media_type;
uint8_t parsed_header[1500];
size_t parsed_header_size, parsed_total_size;
parsed_log.GetRtpHeader(index, &parsed_direction, &parsed_media_type,
parsed_header, &parsed_header_size,
&parsed_total_size);
parsed_log.GetRtpHeader(index, &parsed_direction, parsed_header,
&parsed_header_size, &parsed_total_size);
EXPECT_EQ(direction, parsed_direction);
EXPECT_EQ(media_type, parsed_media_type);
ASSERT_EQ(header_size, parsed_header_size);
EXPECT_EQ(0, std::memcmp(header, parsed_header, header_size));
EXPECT_EQ(total_size, parsed_total_size);
@ -405,7 +385,6 @@ void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
void RtcEventLogTestHelper::VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
PacketDirection direction,
MediaType media_type,
const uint8_t* packet,
size_t total_size) {
const rtclog::Event& event = parsed_log.events_[index];
@ -414,8 +393,6 @@ void RtcEventLogTestHelper::VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
ASSERT_TRUE(rtcp_packet.has_incoming());
EXPECT_EQ(direction == kIncomingPacket, rtcp_packet.incoming());
ASSERT_TRUE(rtcp_packet.has_type());
EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type()));
ASSERT_TRUE(rtcp_packet.has_packet_data());
ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
for (size_t i = 0; i < total_size; i++) {
@ -424,13 +401,11 @@ void RtcEventLogTestHelper::VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
// Check consistency of the parser.
PacketDirection parsed_direction;
MediaType parsed_media_type;
uint8_t parsed_packet[1500];
size_t parsed_total_size;
parsed_log.GetRtcpPacket(index, &parsed_direction, &parsed_media_type,
parsed_packet, &parsed_total_size);
parsed_log.GetRtcpPacket(index, &parsed_direction, parsed_packet,
&parsed_total_size);
EXPECT_EQ(direction, parsed_direction);
EXPECT_EQ(media_type, parsed_media_type);
ASSERT_EQ(total_size, parsed_total_size);
EXPECT_EQ(0, std::memcmp(packet, parsed_packet, total_size));
}

2
webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
View File

@ -35,14 +35,12 @@ class RtcEventLogTestHelper {
static void VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
PacketDirection direction,
MediaType media_type,
const uint8_t* header,
size_t header_size,
size_t total_size);
static void VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
PacketDirection direction,
MediaType media_type,
const uint8_t* packet,
size_t total_size);
static void VerifyPlayoutEvent(const ParsedRtcEventLog& parsed_log,

55
webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
View File

@ -39,38 +39,45 @@ bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
}
std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
while (rtp_packet_index_ < parsed_stream_.GetNumberOfEvents()) {
for (; rtp_packet_index_ < parsed_stream_.GetNumberOfEvents();
rtp_packet_index_++) {
if (parsed_stream_.GetEventType(rtp_packet_index_) ==
ParsedRtcEventLog::RTP_EVENT) {
PacketDirection direction;
MediaType media_type;
size_t header_length;
size_t packet_length;
uint64_t timestamp_us = parsed_stream_.GetTimestamp(rtp_packet_index_);
parsed_stream_.GetRtpHeader(rtp_packet_index_, &direction, &media_type,
nullptr, &header_length, &packet_length);
if (direction == kIncomingPacket && media_type == MediaType::AUDIO) {
uint8_t* packet_header = new uint8_t[header_length];
parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, nullptr,
packet_header, nullptr, nullptr);
std::unique_ptr<Packet> packet(new Packet(
packet_header, header_length, packet_length,
static_cast<double>(timestamp_us) / 1000, *parser_.get()));
if (packet->valid_header()) {
// Check if the packet should not be filtered out.
if (!filter_.test(packet->header().payloadType) &&
!(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
rtp_packet_index_++;
return packet;
}
} else {
std::cout << "Warning: Packet with index " << rtp_packet_index_
<< " has an invalid header and will be ignored."
<< std::endl;
}
parsed_stream_.GetRtpHeader(rtp_packet_index_, &direction, nullptr,
&header_length, &packet_length);
if (direction != kIncomingPacket) {
continue;
}
uint8_t* packet_header = new uint8_t[header_length];
parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, packet_header,
nullptr, nullptr);
std::unique_ptr<Packet> packet(
new Packet(packet_header, header_length, packet_length,
static_cast<double>(timestamp_us) / 1000, *parser_.get()));
if (!packet->valid_header()) {
std::cout << "Warning: Packet with index " << rtp_packet_index_
<< " has an invalid header and will be ignored." << std::endl;
continue;
}
if (parsed_stream_.GetMediaType(packet->header().ssrc, direction) !=
webrtc::ParsedRtcEventLog::MediaType::AUDIO) {
continue;
}
// Check if the packet should not be filtered out.
if (!filter_.test(packet->header().payloadType) &&
!(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
return packet;
}
}
rtp_packet_index_++;
}
return nullptr;
}

1
webrtc/modules/congestion_controller/BUILD.gn
View File

@ -86,6 +86,7 @@ if (rtc_include_tests) {
deps = [
":congestion_controller",
":mock_congestion_controller",
"../../base:rtc_base",
"../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:field_trial",

2
webrtc/modules/congestion_controller/congestion_controller_unittest.cc
View File

@ -9,6 +9,7 @@
*/
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/base/socket.h"
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/congestion_controller/congestion_controller_unittests_helper.h"
@ -16,6 +17,7 @@
#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/gmock.h"

6
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -100,8 +100,7 @@ class PacketContainer : public rtcp::CompoundPacket,
if (transport_->SendRtcp(data, length)) {
bytes_sent_ += length;
if (event_log_) {
event_log_->LogRtcpPacket(kOutgoingPacket, MediaType::ANY, data,
length);
event_log_->LogRtcpPacket(kOutgoingPacket, data, length);
}
}
}
@ -987,8 +986,7 @@ bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
void OnPacketReady(uint8_t* data, size_t length) override {
if (transport_->SendRtcp(data, length)) {
if (event_log_) {
event_log_->LogRtcpPacket(kOutgoingPacket, MediaType::ANY, data,
length);
event_log_->LogRtcpPacket(kOutgoingPacket, data, length);
}
} else {
send_failure_ = true;

4
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -631,8 +631,8 @@ bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {
event_log_->LogRtpHeader(kOutgoingPacket, MediaType::ANY, packet.data(),
packet.size(), pacing_info.probe_cluster_id);
event_log_->LogRtpHeader(kOutgoingPacket, packet.data(), packet.size(),
pacing_info.probe_cluster_id);
}
}
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),

14
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -464,7 +464,7 @@ TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
kSsrc, kSeqNum, _, _, _));
EXPECT_CALL(mock_rtc_event_log_,
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _));
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@ -509,7 +509,7 @@ TEST_P(RtpSenderTest, TrafficSmoothingRetransmits) {
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
kSsrc, kSeqNum, _, _, _));
EXPECT_CALL(mock_rtc_event_log_,
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _));
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@ -563,7 +563,7 @@ TEST_P(RtpSenderTest, SendPadding) {
EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
kSsrc, kSeqNum, _, _, _));
EXPECT_CALL(mock_rtc_event_log_,
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _))
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
.Times(1 + 4 + 1);
uint16_t seq_num = kSeqNum;
@ -764,7 +764,7 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _))
.Times(kNumPayloadSizes);
EXPECT_CALL(mock_rtc_event_log_,
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _))
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
.Times(kNumPayloadSizes);
// Send 10 packets of increasing size.
@ -778,7 +778,7 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
}
EXPECT_CALL(mock_rtc_event_log_,
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _))
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
.Times(::testing::AtLeast(4));
// The amount of padding to send it too small to send a payload packet.
@ -875,7 +875,7 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
.WillOnce(testing::SaveArg<2>(&flexfec_seq_num));
SendGenericPayload();
EXPECT_CALL(mock_rtc_event_log_,
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _))
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
.Times(2);
EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum,
fake_clock_.TimeInMilliseconds(),
@ -923,7 +923,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
rtp_sender_->SetFecParameters(params, params);
EXPECT_CALL(mock_rtc_event_log_,
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _))
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
.Times(2);
SendGenericPayload();
ASSERT_EQ(2, transport_.packets_sent());

12
webrtc/tools/event_log_visualizer/analyzer.cc
View File

@ -373,9 +373,8 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
break;
}
case ParsedRtcEventLog::RTP_EVENT: {
MediaType media_type;
parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
&header_length, &total_length);
parsed_log_.GetRtpHeader(i, &direction, header, &header_length,
&total_length);
// Parse header to get SSRC.
RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
RTPHeader parsed_header;
@ -399,9 +398,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
}
case ParsedRtcEventLog::RTCP_EVENT: {
uint8_t packet[IP_PACKET_SIZE];
MediaType media_type;
parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet,
&total_length);
parsed_log_.GetRtcpPacket(i, &direction, packet, &total_length);
// Currently incoming RTCP packets are logged twice, both for audio and
// video. Only act on one of them. Compare against the previous parsed
// incoming RTCP packet.
@ -905,8 +902,7 @@ void EventLogAnalyzer::CreateTotalBitrateGraph(
for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
if (event_type == ParsedRtcEventLog::RTP_EVENT) {
parsed_log_.GetRtpHeader(i, &direction, nullptr, nullptr, nullptr,
&total_length);
parsed_log_.GetRtpHeader(i, &direction, nullptr, nullptr, &total_length);
if (direction == desired_direction) {
uint64_t timestamp = parsed_log_.GetTimestamp(i);
packets.push_back(TimestampSize(timestamp, total_length));

10
webrtc/voice_engine/channel.cc
View File

@ -101,32 +101,28 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog {
}
void LogRtpHeader(webrtc::PacketDirection direction,
webrtc::MediaType media_type,
const uint8_t* header,
size_t packet_length) override {
LogRtpHeader(direction, media_type, header, packet_length,
PacedPacketInfo::kNotAProbe);
LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe);
}
void LogRtpHeader(webrtc::PacketDirection direction,
webrtc::MediaType media_type,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogRtpHeader(direction, media_type, header, packet_length,
event_log_->LogRtpHeader(direction, header, packet_length,
probe_cluster_id);
}
}
void LogRtcpPacket(webrtc::PacketDirection direction,
webrtc::MediaType media_type,
const uint8_t* packet,
size_t length) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogRtcpPacket(direction, media_type, packet, length);
event_log_->LogRtcpPacket(direction, packet, length);
}
}