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Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets.

Browse Source 
Change LogIncomingRtcpPacket and LogOutgoingRtcpPacket to take ArrayView<uint8_t>.
Split LogSessionAndReadBack into three functions and create class to share state between them.
Split VerifyRtpEvent into one incoming and one outgoing version.

Originally uploaded as https://codereview.webrtc.org/2997973002/

Bug: webrtc:8111
Change-Id: I22bdc35163bef60bc8293679226b19e41e8f49b3
Reviewed-on: https://webrtc-review.googlesource.com/5020
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20063}
This commit is contained in:
Bjorn Terelius
2017-09-29 21:01:42 +02:00
committed by Commit Bot
parent 5117b04787
commit 440216fcf3
12 changed files with 685 additions and 354 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
logging/BUILD.gn
View File

@ -31,6 +31,7 @@ rtc_source_set("rtc_event_log_api") {
]
deps = [
"..:webrtc_common",
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../call:video_stream_api",
"../rtc_base:rtc_base_approved",

23
logging/rtc_event_log/mock/mock_rtc_event_log.h
View File

@ -16,6 +16,8 @@
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "test/gmock.h"
namespace webrtc {
@ -42,21 +44,16 @@ class MockRtcEventLog : public RtcEventLog {
MOCK_METHOD1(LogAudioSendStreamConfig,
void(const rtclog::StreamConfig& config));
MOCK_METHOD3(LogRtpHeader,
void(PacketDirection direction,
const uint8_t* header,
size_t packet_length));
MOCK_METHOD1(LogIncomingRtpHeader, void(const RtpPacketReceived& packet));
MOCK_METHOD4(LogRtpHeader,
void(PacketDirection direction,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id));
MOCK_METHOD2(LogOutgoingRtpHeader,
void(const RtpPacketToSend& packet, int probe_cluster_id));
MOCK_METHOD3(LogRtcpPacket,
void(PacketDirection direction,
const uint8_t* packet,
size_t length));
MOCK_METHOD1(LogIncomingRtcpPacket,
void(rtc::ArrayView<const uint8_t> packet));
MOCK_METHOD1(LogOutgoingRtcpPacket,
void(rtc::ArrayView<const uint8_t> packet));
MOCK_METHOD1(LogAudioPlayout, void(uint32_t ssrc));

33
logging/rtc_event_log/rtc_event_log.cc
View File

@ -33,6 +33,8 @@
#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/event.h"
@ -112,16 +114,27 @@ class RtcEventLogImpl final : public RtcEventLog {
void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override;
void LogAudioReceiveStreamConfig(const rtclog::StreamConfig& config) override;
void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override;
// TODO(terelius): This can be removed as soon as the interface has been
// updated.
void LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length) override;
// TODO(terelius): This can be made private, non-virtual as soon as the
// interface has been updated.
void LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) override;
void LogIncomingRtpHeader(const RtpPacketReceived& packet) override;
void LogOutgoingRtpHeader(const RtpPacketToSend& packet,
int probe_cluster_id) override;
// TODO(terelius): This can be made private, non-virtual as soon as the
// interface has been updated.
void LogRtcpPacket(PacketDirection direction,
const uint8_t* packet,
size_t length) override;
void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override;
void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override;
void LogAudioPlayout(uint32_t ssrc) override;
void LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
@ -418,6 +431,16 @@ void RtcEventLogImpl::LogAudioSendStreamConfig(
StoreEvent(std::move(event));
}
void RtcEventLogImpl::LogIncomingRtpHeader(const RtpPacketReceived& packet) {
LogRtpHeader(kIncomingPacket, packet.data(), packet.size(),
PacedPacketInfo::kNotAProbe);
}
void RtcEventLogImpl::LogOutgoingRtpHeader(const RtpPacketToSend& packet,
int probe_cluster_id) {
LogRtpHeader(kOutgoingPacket, packet.data(), packet.size(), probe_cluster_id);
}
void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length) {
@ -455,6 +478,16 @@ void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
StoreEvent(std::move(rtp_event));
}
void RtcEventLogImpl::LogIncomingRtcpPacket(
rtc::ArrayView<const uint8_t> packet) {
LogRtcpPacket(kIncomingPacket, packet.data(), packet.size());
}
void RtcEventLogImpl::LogOutgoingRtcpPacket(
rtc::ArrayView<const uint8_t> packet) {
LogRtcpPacket(kOutgoingPacket, packet.data(), packet.size());
}
void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
const uint8_t* packet,
size_t length) {

58
logging/rtc_event_log/rtc_event_log.h
View File

@ -15,6 +15,7 @@
#include <string>
#include <vector>
#include "api/array_view.h"
// TODO(eladalon): Get rid of this later in the CL-stack.
#include "api/rtpparameters.h"
#include "common_types.h" // NOLINT(build/include)
@ -30,6 +31,8 @@ struct StreamConfig;
class Clock;
struct AudioEncoderRuntimeConfig;
class RtpPacketReceived;
class RtpPacketToSend;
enum class MediaType;
enum class BandwidthUsage;
@ -49,7 +52,7 @@ class RtcEventLog {
static std::unique_ptr<RtcEventLog> Create();
// TODO(nisse): webrtc::Clock is deprecated. Delete this method and
// above forward declaration of Clock when
// webrtc/system_wrappers/include/clock.h is deleted.
// system_wrappers/include/clock.h is deleted.
static std::unique_ptr<RtcEventLog> Create(const Clock* clock) {
return Create();
}
@ -98,23 +101,33 @@ class RtcEventLog {
// 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
RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length) {}
RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) {}
// Logs the header of an incoming RTP packet. |packet_length|
// is the total length of the packet, including both header and payload.
virtual void LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length) = 0;
virtual void LogIncomingRtpHeader(const RtpPacketReceived& packet) = 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,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) = 0;
// Logs the header of an incoming RTP packet. |packet_length|
// is the total length of the packet, including both header and payload.
virtual void LogOutgoingRtpHeader(const RtpPacketToSend& packet,
int probe_cluster_id) = 0;
// Logs an incoming or outgoing RTCP packet.
virtual void LogRtcpPacket(PacketDirection direction,
const uint8_t* packet,
size_t length) = 0;
RTC_DEPRECATED virtual void LogRtcpPacket(PacketDirection direction,
const uint8_t* header,
size_t packet_length) {}
// Logs an incoming RTCP packet.
virtual void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0;
// Logs an outgoing RTCP packet.
virtual void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0;
// Logs an audio playout event.
virtual void LogAudioPlayout(uint32_t ssrc) = 0;
@ -164,16 +177,11 @@ class RtcEventLogNullImpl : public RtcEventLog {
void LogAudioReceiveStreamConfig(
const rtclog::StreamConfig& config) override {}
void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {}
void LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length) override {}
void LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) override {}
void LogRtcpPacket(PacketDirection direction,
const uint8_t* packet,
size_t length) override {}
void LogIncomingRtpHeader(const RtpPacketReceived& packet) override {}
void LogOutgoingRtpHeader(const RtpPacketToSend& packet,
int probe_cluster_id) override {}
void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {}
void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {}
void LogAudioPlayout(uint32_t ssrc) override {}
void LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,

764
logging/rtc_event_log/rtc_event_log_unittest.cc
View File

@ -10,6 +10,7 @@
#include <map>
#include <memory>
#include <ostream>
#include <string>
#include <utility>
#include <vector>
@ -25,10 +26,12 @@
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/fakeclock.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/random.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@ -44,72 +47,105 @@ namespace webrtc {
namespace {
const uint8_t kTransmissionTimeOffsetExtensionId = 1;
const uint8_t kAbsoluteSendTimeExtensionId = 14;
const uint8_t kTransportSequenceNumberExtensionId = 13;
const uint8_t kAudioLevelExtensionId = 9;
const uint8_t kVideoRotationExtensionId = 5;
const uint8_t kExtensionIds[] = {
kTransmissionTimeOffsetExtensionId, kAbsoluteSendTimeExtensionId,
kTransportSequenceNumberExtensionId, kAudioLevelExtensionId,
kVideoRotationExtensionId};
const RTPExtensionType kExtensionTypes[] = {
RTPExtensionType::kRtpExtensionTransmissionTimeOffset,
RTPExtensionType::kRtpExtensionAudioLevel,
RTPExtensionType::kRtpExtensionAbsoluteSendTime,
RTPExtensionType::kRtpExtensionVideoRotation,
RTPExtensionType::kRtpExtensionTransportSequenceNumber};
RTPExtensionType::kRtpExtensionTransportSequenceNumber,
RTPExtensionType::kRtpExtensionAudioLevel,
RTPExtensionType::kRtpExtensionVideoRotation};
const char* kExtensionNames[] = {
RtpExtension::kTimestampOffsetUri, RtpExtension::kAudioLevelUri,
RtpExtension::kAbsSendTimeUri, RtpExtension::kVideoRotationUri,
RtpExtension::kTransportSequenceNumberUri};
RtpExtension::kTimestampOffsetUri, RtpExtension::kAbsSendTimeUri,
RtpExtension::kTransportSequenceNumberUri, RtpExtension::kAudioLevelUri,
RtpExtension::kVideoRotationUri};
const size_t kNumExtensions = 5;
void PrintActualEvents(const ParsedRtcEventLog& parsed_log) {
std::map<int, size_t> actual_event_counts;
for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) {
actual_event_counts[parsed_log.GetEventType(i)]++;
}
printf("Actual events: ");
for (auto kv : actual_event_counts) {
printf("%d_count = %zu, ", kv.first, kv.second);
}
printf("\n");
for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) {
printf("%4d ", parsed_log.GetEventType(i));
}
printf("\n");
}
struct BweLossEvent {
int32_t bitrate_bps;
uint8_t fraction_loss;
int32_t total_packets;
};
void PrintExpectedEvents(size_t rtp_count,
size_t rtcp_count,
size_t playout_count,
size_t bwe_loss_count) {
printf(
"Expected events: rtp_count = %zu, rtcp_count = %zu,"
"playout_count = %zu, bwe_loss_count = %zu\n",
rtp_count, rtcp_count, playout_count, bwe_loss_count);
size_t rtcp_index = 1, playout_index = 1, bwe_loss_index = 1;
printf("strt cfg cfg ");
for (size_t i = 1; i <= rtp_count; i++) {
printf(" rtp ");
if (i * rtcp_count >= rtcp_index * rtp_count) {
printf("rtcp ");
rtcp_index++;
}
if (i * playout_count >= playout_index * rtp_count) {
printf("play ");
playout_index++;
}
if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
printf("loss ");
bwe_loss_index++;
}
}
printf("end \n");
}
// TODO(terelius): Merge with event type in parser once updated?
enum class EventType {
kIncomingRtp,
kOutgoingRtp,
kIncomingRtcp,
kOutgoingRtcp,
kAudioPlayout,
kBweLossUpdate,
kBweDelayUpdate,
kVideoRecvConfig,
kVideoSendConfig,
kAudioRecvConfig,
kAudioSendConfig,
kAudioNetworkAdaptation,
kBweProbeClusterCreated,
kBweProbeResult,
};
const std::map<EventType, std::string> event_type_to_string(
{{EventType::kIncomingRtp, "RTP(in)"},
{EventType::kOutgoingRtp, "RTP(out)"},
{EventType::kIncomingRtcp, "RTCP(in)"},
{EventType::kOutgoingRtcp, "RTCP(out)"},
{EventType::kAudioPlayout, "PLAYOUT"},
{EventType::kBweLossUpdate, "BWE_LOSS"},
{EventType::kBweDelayUpdate, "BWE_DELAY"},
{EventType::kVideoRecvConfig, "VIDEO_RECV_CONFIG"},
{EventType::kVideoSendConfig, "VIDEO_SEND_CONFIG"},
{EventType::kAudioRecvConfig, "AUDIO_RECV_CONFIG"},
{EventType::kAudioSendConfig, "AUDIO_SEND_CONFIG"},
{EventType::kAudioNetworkAdaptation, "AUDIO_NETWORK_ADAPTATION"},
{EventType::kBweProbeClusterCreated, "BWE_PROBE_CREATED"},
{EventType::kBweProbeResult, "BWE_PROBE_RESULT"}});
const std::map<ParsedRtcEventLog::EventType, std::string>
parsed_event_type_to_string(
{{ParsedRtcEventLog::EventType::UNKNOWN_EVENT, "UNKNOWN_EVENT"},
{ParsedRtcEventLog::EventType::LOG_START, "LOG_START"},
{ParsedRtcEventLog::EventType::LOG_END, "LOG_END"},
{ParsedRtcEventLog::EventType::RTP_EVENT, "RTP"},
{ParsedRtcEventLog::EventType::RTCP_EVENT, "RTCP"},
{ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT, "AUDIO_PLAYOUT"},
{ParsedRtcEventLog::EventType::LOSS_BASED_BWE_UPDATE,
"LOSS_BASED_BWE_UPDATE"},
{ParsedRtcEventLog::EventType::DELAY_BASED_BWE_UPDATE,
"DELAY_BASED_BWE_UPDATE"},
{ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT,
"VIDEO_RECV_CONFIG"},
{ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT,
"VIDEO_SEND_CONFIG"},
{ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT,
"AUDIO_RECV_CONFIG"},
{ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT,
"AUDIO_SEND_CONFIG"},
{ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT,
"AUDIO_NETWORK_ADAPTATION"},
{ParsedRtcEventLog::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT,
"BWE_PROBE_CREATED"},
{ParsedRtcEventLog::EventType::BWE_PROBE_RESULT_EVENT,
"BWE_PROBE_RESULT"}});
} // namespace
/*
* Bit number i of extension_bitvector is set to indicate the
* presence of extension number i from kExtensionTypes / kExtensionNames.
* The least significant bit extension_bitvector has number 0.
*/
RtpPacketToSend GenerateRtpPacket(const RtpHeaderExtensionMap* extensions,
uint32_t csrcs_count,
size_t packet_size,
Random* prng) {
void PrintActualEvents(const ParsedRtcEventLog& parsed_log,
std::ostream& stream);
RtpPacketToSend GenerateOutgoingRtpPacket(
const RtpHeaderExtensionMap* extensions,
uint32_t csrcs_count,
size_t packet_size,
Random* prng) {
RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
std::vector<uint32_t> csrcs;
@ -139,6 +175,18 @@ RtpPacketToSend GenerateRtpPacket(const RtpHeaderExtensionMap* extensions,
return rtp_packet;
}
RtpPacketReceived GenerateIncomingRtpPacket(
const RtpHeaderExtensionMap* extensions,
uint32_t csrcs_count,
size_t packet_size,
Random* prng) {
RtpPacketToSend packet_out =
GenerateOutgoingRtpPacket(extensions, csrcs_count, packet_size, prng);
RtpPacketReceived packet_in(extensions);
packet_in.Parse(packet_out.data(), packet_out.size());
return packet_in;
}
rtc::Buffer GenerateRtcpPacket(Random* prng) {
rtcp::ReportBlock report_block;
report_block.SetMediaSsrc(prng->Rand<uint32_t>()); // Remote SSRC.
@ -153,7 +201,7 @@ rtc::Buffer GenerateRtcpPacket(Random* prng) {
return sender_report.Build();
}
void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
void GenerateVideoReceiveConfig(const RtpHeaderExtensionMap& extensions,
rtclog::StreamConfig* config,
Random* prng) {
// Add SSRCs for the stream.
@ -168,14 +216,14 @@ void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
prng->Rand(1, 127), prng->Rand(1, 127));
// Add header extensions.
for (unsigned i = 0; i < kNumExtensions; i++) {
if (extensions_bitvector & (1u << i)) {
config->rtp_extensions.emplace_back(kExtensionNames[i],
prng->Rand<int>());
uint8_t id = extensions.GetId(kExtensionTypes[i]);
if (id != RtpHeaderExtensionMap::kInvalidId) {
config->rtp_extensions.emplace_back(kExtensionNames[i], id);
}
}
}
void GenerateVideoSendConfig(uint32_t extensions_bitvector,
void GenerateVideoSendConfig(const RtpHeaderExtensionMap& extensions,
rtclog::StreamConfig* config,
Random* prng) {
config->codecs.emplace_back(prng->Rand<bool>() ? "VP8" : "H264",
@ -184,14 +232,14 @@ void GenerateVideoSendConfig(uint32_t extensions_bitvector,
config->rtx_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(
RtpExtension(kExtensionNames[i], prng->Rand<int>()));
uint8_t id = extensions.GetId(kExtensionTypes[i]);
if (id != RtpHeaderExtensionMap::kInvalidId) {
config->rtp_extensions.emplace_back(kExtensionNames[i], id);
}
}
}
void GenerateAudioReceiveConfig(uint32_t extensions_bitvector,
void GenerateAudioReceiveConfig(const RtpHeaderExtensionMap& extensions,
rtclog::StreamConfig* config,
Random* prng) {
// Add SSRCs for the stream.
@ -199,28 +247,36 @@ void GenerateAudioReceiveConfig(uint32_t extensions_bitvector,
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(
RtpExtension(kExtensionNames[i], prng->Rand<int>()));
uint8_t id = extensions.GetId(kExtensionTypes[i]);
if (id != RtpHeaderExtensionMap::kInvalidId) {
config->rtp_extensions.emplace_back(kExtensionNames[i], id);
}
}
}
void GenerateAudioSendConfig(uint32_t extensions_bitvector,
void GenerateAudioSendConfig(const RtpHeaderExtensionMap& extensions,
rtclog::StreamConfig* config,
Random* prng) {
// Add SSRC to the stream.
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(
RtpExtension(kExtensionNames[i], prng->Rand<int>()));
uint8_t id = extensions.GetId(kExtensionTypes[i]);
if (id != RtpHeaderExtensionMap::kInvalidId) {
config->rtp_extensions.emplace_back(kExtensionNames[i], id);
}
}
}
void GenerateAudioNetworkAdaptation(uint32_t extensions_bitvector,
BweLossEvent GenerateBweLossEvent(Random* prng) {
BweLossEvent loss_event;
loss_event.bitrate_bps = prng->Rand(6000, 10000000);
loss_event.fraction_loss = prng->Rand<uint8_t>();
loss_event.total_packets = prng->Rand(1, 1000);
return loss_event;
}
void GenerateAudioNetworkAdaptation(const RtpHeaderExtensionMap& extensions,
AudioEncoderRuntimeConfig* config,
Random* prng) {
config->bitrate_bps = rtc::Optional<int>(prng->Rand(0, 3000000));
@ -232,201 +288,414 @@ void GenerateAudioNetworkAdaptation(uint32_t extensions_bitvector,
rtc::Optional<float>(prng->Rand<float>());
}
// Test for the RtcEventLog class. Dumps some RTP packets and other events
// to disk, then reads them back to see if they match.
void LogSessionAndReadBack(size_t rtp_count,
size_t rtcp_count,
size_t playout_count,
size_t bwe_loss_count,
uint32_t extensions_bitvector,
uint32_t csrcs_count,
unsigned int random_seed) {
ASSERT_LE(rtcp_count, rtp_count);
ASSERT_LE(playout_count, rtp_count);
ASSERT_LE(bwe_loss_count, rtp_count);
std::vector<RtpPacketToSend> rtp_packets;
std::vector<rtc::Buffer> rtcp_packets;
class RtcEventLogSessionDescription {
public:
explicit RtcEventLogSessionDescription(unsigned int random_seed)
: prng(random_seed) {}
void GenerateSessionDescription(size_t incoming_rtp_count,
size_t outgoing_rtp_count,
size_t incoming_rtcp_count,
size_t outgoing_rtcp_count,
size_t playout_count,
size_t bwe_loss_count,
size_t bwe_delay_count,
const RtpHeaderExtensionMap& extensions,
uint32_t csrcs_count);
void WriteSession();
void ReadAndVerifySession();
void PrintExpectedEvents(std::ostream& stream);
private:
std::vector<RtpPacketReceived> incoming_rtp_packets;
std::vector<RtpPacketToSend> outgoing_rtp_packets;
std::vector<rtc::Buffer> incoming_rtcp_packets;
std::vector<rtc::Buffer> outgoing_rtcp_packets;
std::vector<uint32_t> playout_ssrcs;
std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates;
std::vector<BweLossEvent> bwe_loss_updates;
std::vector<std::pair<int32_t, BandwidthUsage> > bwe_delay_updates;
std::vector<rtclog::StreamConfig> receiver_configs;
std::vector<rtclog::StreamConfig> sender_configs;
std::vector<EventType> event_types;
Random prng;
};
rtclog::StreamConfig receiver_config;
rtclog::StreamConfig sender_config;
void RtcEventLogSessionDescription::GenerateSessionDescription(
size_t incoming_rtp_count,
size_t outgoing_rtp_count,
size_t incoming_rtcp_count,
size_t outgoing_rtcp_count,
size_t playout_count,
size_t bwe_loss_count,
size_t bwe_delay_count,
const RtpHeaderExtensionMap& extensions,
uint32_t csrcs_count) {
// Create configuration for the video receive stream.
receiver_configs.push_back(rtclog::StreamConfig());
GenerateVideoReceiveConfig(extensions, &receiver_configs.back(), &prng);
event_types.push_back(EventType::kVideoRecvConfig);
Random prng(random_seed);
// Create configuration for the video send stream.
sender_configs.push_back(rtclog::StreamConfig());
GenerateVideoSendConfig(extensions, &sender_configs.back(), &prng);
event_types.push_back(EventType::kVideoSendConfig);
const size_t config_count = 2;
// Initialize rtp header extensions to be used in generated rtp packets.
RtpHeaderExtensionMap extensions;
for (unsigned i = 0; i < kNumExtensions; i++) {
if (extensions_bitvector & (1u << i)) {
extensions.Register(kExtensionTypes[i], i + 1);
}
}
// Create rtp_count RTP packets containing random data.
for (size_t i = 0; i < rtp_count; i++) {
// Create incoming and outgoing RTP packets containing random data.
for (size_t i = 0; i < incoming_rtp_count; i++) {
size_t packet_size = prng.Rand(1000, 1100);
rtp_packets.push_back(
GenerateRtpPacket(&extensions, csrcs_count, packet_size, &prng));
incoming_rtp_packets.push_back(GenerateIncomingRtpPacket(
&extensions, csrcs_count, packet_size, &prng));
event_types.push_back(EventType::kIncomingRtp);
}
// Create rtcp_count RTCP packets containing random data.
for (size_t i = 0; i < rtcp_count; i++) {
rtcp_packets.push_back(GenerateRtcpPacket(&prng));
for (size_t i = 0; i < outgoing_rtp_count; i++) {
size_t packet_size = prng.Rand(1000, 1100);
outgoing_rtp_packets.push_back(GenerateOutgoingRtpPacket(
&extensions, csrcs_count, packet_size, &prng));
event_types.push_back(EventType::kOutgoingRtp);
}
// Create playout_count random SSRCs to use when logging AudioPlayout events.
// Create incoming and outgoing RTCP packets containing random data.
for (size_t i = 0; i < incoming_rtcp_count; i++) {
incoming_rtcp_packets.push_back(GenerateRtcpPacket(&prng));
event_types.push_back(EventType::kIncomingRtcp);
}
for (size_t i = 0; i < outgoing_rtcp_count; i++) {
outgoing_rtcp_packets.push_back(GenerateRtcpPacket(&prng));
event_types.push_back(EventType::kOutgoingRtcp);
}
// Create random SSRCs to use when logging AudioPlayout events.
for (size_t i = 0; i < playout_count; i++) {
playout_ssrcs.push_back(prng.Rand<uint32_t>());
event_types.push_back(EventType::kAudioPlayout);
}
// Create bwe_loss_count random bitrate updates for LossBasedBwe.
// Create random bitrate updates for LossBasedBwe.
for (size_t i = 0; i < bwe_loss_count; i++) {
bwe_loss_updates.push_back(
std::make_pair(prng.Rand<int32_t>(), prng.Rand<uint8_t>()));
bwe_loss_updates.push_back(GenerateBweLossEvent(&prng));
event_types.push_back(EventType::kBweLossUpdate);
}
// Create random bitrate updates for DelayBasedBwe.
for (size_t i = 0; i < bwe_delay_count; i++) {
bwe_delay_updates.push_back(std::make_pair(
prng.Rand(6000, 10000000), prng.Rand<bool>()
? BandwidthUsage::kBwOverusing
: BandwidthUsage::kBwUnderusing));
event_types.push_back(EventType::kBweDelayUpdate);
}
// Create configurations for the video streams.
GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng);
const int config_count = 2;
// Order the events randomly. The configurations are stored in a separate
// buffer, so they might be written before any othe events. Hence, we can't
// mix the config events with other events.
for (size_t i = config_count; i < event_types.size(); i++) {
size_t other = prng.Rand(static_cast<uint32_t>(i),
static_cast<uint32_t>(event_types.size() - 1));
RTC_CHECK(i <= other && other < event_types.size());
std::swap(event_types[i], event_types[other]);
}
}
void RtcEventLogSessionDescription::WriteSession() {
// Find the name of the current test, in order to use it as a temporary
// filename.
auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
const std::string temp_filename =
test::OutputPath() + test_info->test_case_name() + test_info->name();
rtc::ScopedFakeClock fake_clock;
fake_clock.SetTimeMicros(prng.Rand<uint32_t>());
// When log_dumper goes out of scope, it causes the log file to be flushed
// to disk.
{
rtc::ScopedFakeClock fake_clock;
fake_clock.SetTimeMicros(prng.Rand<uint32_t>());
std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
log_dumper->LogVideoReceiveStreamConfig(receiver_config);
std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
size_t incoming_rtp_written = 0;
size_t outgoing_rtp_written = 0;
size_t incoming_rtcp_written = 0;
size_t outgoing_rtcp_written = 0;
size_t playouts_written = 0;
size_t bwe_loss_written = 0;
size_t bwe_delay_written = 0;
size_t recv_configs_written = 0;
size_t send_configs_written = 0;
for (size_t i = 0; i < event_types.size(); i++) {
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->LogVideoSendStreamConfig(sender_config);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
size_t rtcp_index = 1;
size_t playout_index = 1;
size_t bwe_loss_index = 1;
for (size_t i = 1; i <= rtp_count; i++) {
log_dumper->LogRtpHeader(
(i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
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_packets[rtcp_index - 1].data(),
rtcp_packets[rtcp_index - 1].size());
rtcp_index++;
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
}
if (i * playout_count >= playout_index * rtp_count) {
log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]);
playout_index++;
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
}
if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
if (i == event_types.size() / 2)
log_dumper->StartLogging(temp_filename, 10000000);
switch (event_types[i]) {
case EventType::kIncomingRtp:
RTC_CHECK(incoming_rtp_written < incoming_rtp_packets.size());
log_dumper->LogIncomingRtpHeader(
incoming_rtp_packets[incoming_rtp_written++]);
break;
case EventType::kOutgoingRtp:
RTC_CHECK(outgoing_rtp_written < outgoing_rtp_packets.size());
log_dumper->LogOutgoingRtpHeader(
outgoing_rtp_packets[outgoing_rtp_written++],
PacedPacketInfo::kNotAProbe);
break;
case EventType::kIncomingRtcp:
RTC_CHECK(incoming_rtcp_written < incoming_rtcp_packets.size());
log_dumper->LogIncomingRtcpPacket(
incoming_rtcp_packets[incoming_rtcp_written++]);
break;
case EventType::kOutgoingRtcp:
RTC_CHECK(outgoing_rtcp_written < outgoing_rtcp_packets.size());
log_dumper->LogOutgoingRtcpPacket(
outgoing_rtcp_packets[outgoing_rtcp_written++]);
break;
case EventType::kAudioPlayout:
RTC_CHECK(playouts_written < playout_ssrcs.size());
log_dumper->LogAudioPlayout(playout_ssrcs[playouts_written++]);
break;
case EventType::kBweLossUpdate:
RTC_CHECK(bwe_loss_written < bwe_loss_updates.size());
log_dumper->LogLossBasedBweUpdate(
bwe_loss_updates[bwe_loss_index - 1].first,
bwe_loss_updates[bwe_loss_index - 1].second, i);
bwe_loss_index++;
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
}
if (i == rtp_count / 2) {
log_dumper->StartLogging(temp_filename, 10000000);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
}
bwe_loss_updates[bwe_loss_written].bitrate_bps,
bwe_loss_updates[bwe_loss_written].fraction_loss,
bwe_loss_updates[bwe_loss_written].total_packets);
bwe_loss_written++;
break;
case EventType::kBweDelayUpdate:
RTC_CHECK(bwe_delay_written < bwe_delay_updates.size());
log_dumper->LogDelayBasedBweUpdate(
bwe_delay_updates[bwe_delay_written].first,
bwe_delay_updates[bwe_delay_written].second);
bwe_delay_written++;
break;
case EventType::kVideoRecvConfig:
RTC_CHECK(recv_configs_written < receiver_configs.size());
log_dumper->LogVideoReceiveStreamConfig(
receiver_configs[recv_configs_written++]);
break;
case EventType::kVideoSendConfig:
RTC_CHECK(send_configs_written < sender_configs.size());
log_dumper->LogVideoSendStreamConfig(
sender_configs[send_configs_written++]);
break;
case EventType::kAudioRecvConfig:
// Not implemented
RTC_NOTREACHED();
break;
case EventType::kAudioSendConfig:
// Not implemented
RTC_NOTREACHED();
break;
case EventType::kAudioNetworkAdaptation:
// Not implemented
RTC_NOTREACHED();
break;
case EventType::kBweProbeClusterCreated:
// Not implemented
RTC_NOTREACHED();
break;
case EventType::kBweProbeResult:
// Not implemented
RTC_NOTREACHED();
break;
}
log_dumper->StopLogging();
}
log_dumper->StopLogging();
}
// Read the file and verify that what we read back from the event log is the
// same as what we wrote down.
void RtcEventLogSessionDescription::ReadAndVerifySession() {
// Find the name of the current test, in order to use it as a temporary
// filename.
auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
const std::string temp_filename =
test::OutputPath() + test_info->test_case_name() + test_info->name();
// Read the generated file from disk.
ParsedRtcEventLog parsed_log;
ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
EXPECT_GE(1000u, event_types.size() +
2); // The events must fit in the message queue.
EXPECT_EQ(event_types.size() + 2, parsed_log.GetNumberOfEvents());
size_t incoming_rtp_read = 0;
size_t outgoing_rtp_read = 0;
size_t incoming_rtcp_read = 0;
size_t outgoing_rtcp_read = 0;
size_t playouts_read = 0;
size_t bwe_loss_read = 0;
size_t bwe_delay_read = 0;
size_t recv_configs_read = 0;
size_t send_configs_read = 0;
// Verify that what we read back from the event log is the same as
// what we wrote down. For RTCP we log the full packets, but for
// RTP we should only log the header.
const size_t event_count = config_count + playout_count + bwe_loss_count +
rtcp_count + rtp_count + 2;
EXPECT_GE(1000u, event_count); // The events must fit in the message queue.
EXPECT_EQ(event_count, parsed_log.GetNumberOfEvents());
if (event_count != parsed_log.GetNumberOfEvents()) {
// Print the expected and actual event types for easier debugging.
PrintActualEvents(parsed_log);
PrintExpectedEvents(rtp_count, rtcp_count, playout_count, bwe_loss_count);
}
RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
RtcEventLogTestHelper::VerifyVideoReceiveStreamConfig(parsed_log, 1,
receiver_config);
RtcEventLogTestHelper::VerifyVideoSendStreamConfig(parsed_log, 2,
sender_config);
size_t event_index = config_count + 1;
size_t rtcp_index = 1;
size_t playout_index = 1;
size_t bwe_loss_index = 1;
for (size_t i = 1; i <= rtp_count; i++) {
RtcEventLogTestHelper::VerifyRtpEvent(
parsed_log, event_index,
(i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
rtp_packets[i - 1].data(), rtp_packets[i - 1].headers_size(),
rtp_packets[i - 1].size());
event_index++;
if (i * rtcp_count >= rtcp_index * rtp_count) {
RtcEventLogTestHelper::VerifyRtcpEvent(
parsed_log, event_index,
rtcp_index % 2 == 0 ? kIncomingPacket : kOutgoingPacket,
rtcp_packets[rtcp_index - 1].data(),
rtcp_packets[rtcp_index - 1].size());
event_index++;
rtcp_index++;
}
if (i * playout_count >= playout_index * rtp_count) {
RtcEventLogTestHelper::VerifyPlayoutEvent(
parsed_log, event_index, playout_ssrcs[playout_index - 1]);
event_index++;
playout_index++;
}
if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
RtcEventLogTestHelper::VerifyBweLossEvent(
parsed_log, event_index, bwe_loss_updates[bwe_loss_index - 1].first,
bwe_loss_updates[bwe_loss_index - 1].second, i);
event_index++;
bwe_loss_index++;
for (size_t i = 0; i < event_types.size(); i++) {
switch (event_types[i]) {
case EventType::kIncomingRtp:
RTC_CHECK(incoming_rtp_read < incoming_rtp_packets.size());
RtcEventLogTestHelper::VerifyIncomingRtpEvent(
parsed_log, i + 1, incoming_rtp_packets[incoming_rtp_read++]);
break;
case EventType::kOutgoingRtp:
RTC_CHECK(outgoing_rtp_read < outgoing_rtp_packets.size());
RtcEventLogTestHelper::VerifyOutgoingRtpEvent(
parsed_log, i + 1, outgoing_rtp_packets[outgoing_rtp_read++]);
break;
case EventType::kIncomingRtcp:
RTC_CHECK(incoming_rtcp_read < incoming_rtcp_packets.size());
RtcEventLogTestHelper::VerifyRtcpEvent(
parsed_log, i + 1, kIncomingPacket,
incoming_rtcp_packets[incoming_rtcp_read].data(),
incoming_rtcp_packets[incoming_rtcp_read].size());
incoming_rtcp_read++;
break;
case EventType::kOutgoingRtcp:
RTC_CHECK(outgoing_rtcp_read < outgoing_rtcp_packets.size());
RtcEventLogTestHelper::VerifyRtcpEvent(
parsed_log, i + 1, kOutgoingPacket,
outgoing_rtcp_packets[outgoing_rtcp_read].data(),
outgoing_rtcp_packets[outgoing_rtcp_read].size());
outgoing_rtcp_read++;
break;
case EventType::kAudioPlayout:
RTC_CHECK(playouts_read < playout_ssrcs.size());
RtcEventLogTestHelper::VerifyPlayoutEvent(
parsed_log, i + 1, playout_ssrcs[playouts_read++]);
break;
case EventType::kBweLossUpdate:
RTC_CHECK(bwe_loss_read < bwe_loss_updates.size());
RtcEventLogTestHelper::VerifyBweLossEvent(
parsed_log, i + 1, bwe_loss_updates[bwe_loss_read].bitrate_bps,
bwe_loss_updates[bwe_loss_read].fraction_loss,
bwe_loss_updates[bwe_loss_read].total_packets);
bwe_loss_read++;
break;
case EventType::kBweDelayUpdate:
RTC_CHECK(bwe_delay_read < bwe_delay_updates.size());
RtcEventLogTestHelper::VerifyBweDelayEvent(
parsed_log, i + 1, bwe_delay_updates[bwe_delay_read].first,
bwe_delay_updates[bwe_delay_read].second);
bwe_delay_read++;
break;
case EventType::kVideoRecvConfig:
RTC_CHECK(recv_configs_read < receiver_configs.size());
RtcEventLogTestHelper::VerifyVideoReceiveStreamConfig(
parsed_log, i + 1, receiver_configs[recv_configs_read++]);
break;
case EventType::kVideoSendConfig:
RTC_CHECK(send_configs_read < sender_configs.size());
RtcEventLogTestHelper::VerifyVideoSendStreamConfig(
parsed_log, i + 1, sender_configs[send_configs_read++]);
break;
case EventType::kAudioRecvConfig:
// Not implemented
RTC_NOTREACHED();
break;
case EventType::kAudioSendConfig:
// Not implemented
RTC_NOTREACHED();
break;
case EventType::kAudioNetworkAdaptation:
// Not implemented
RTC_NOTREACHED();
break;
case EventType::kBweProbeClusterCreated:
// Not implemented
RTC_NOTREACHED();
break;
case EventType::kBweProbeResult:
// Not implemented
RTC_NOTREACHED();
break;
}
}
RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log,
parsed_log.GetNumberOfEvents() - 1);
// Clean up temporary file - can be pretty slow.
remove(temp_filename.c_str());
}
TEST(RtcEventLogTest, LogSessionAndReadBack) {
// Log 5 RTP, 2 RTCP, 0 playout events and 0 BWE events
// with no header extensions or CSRCS.
LogSessionAndReadBack(5, 2, 0, 0, 0, 0, 321);
// Enable AbsSendTime and TransportSequenceNumbers.
uint32_t extensions = 0;
for (uint32_t i = 0; i < kNumExtensions; i++) {
if (kExtensionTypes[i] == RTPExtensionType::kRtpExtensionAbsoluteSendTime ||
kExtensionTypes[i] ==
RTPExtensionType::kRtpExtensionTransportSequenceNumber) {
extensions |= 1u << i;
}
void RtcEventLogSessionDescription::PrintExpectedEvents(std::ostream& stream) {
for (size_t i = 0; i < event_types.size(); i++) {
auto it = event_type_to_string.find(event_types[i]);
RTC_CHECK(it != event_type_to_string.end());
stream << it->second << " ";
}
LogSessionAndReadBack(8, 2, 0, 0, extensions, 0, 3141592653u);
stream << std::endl;
}
extensions = (1u << kNumExtensions) - 1; // Enable all header extensions.
LogSessionAndReadBack(9, 2, 3, 2, extensions, 2, 2718281828u);
void PrintActualEvents(const ParsedRtcEventLog& parsed_log,
std::ostream& stream) {
for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) {
auto it = parsed_event_type_to_string.find(parsed_log.GetEventType(i));
RTC_CHECK(it != parsed_event_type_to_string.end());
stream << it->second << " ";
}
stream << std::endl;
}
TEST(RtcEventLogTest, LogSessionAndReadBack) {
RtpHeaderExtensionMap extensions;
RtcEventLogSessionDescription session(321 /*Random seed*/);
session.GenerateSessionDescription(3, // Number of incoming RTP packets.
2, // Number of outgoing RTP packets.
1, // Number of incoming RTCP packets.
1, // Number of outgoing RTCP packets.
0, // Number of playout events.
0, // Number of BWE loss events.
0, // Number of BWE delay events.
extensions, // No extensions.
0); // Number of contributing sources.
session.WriteSession();
session.ReadAndVerifySession();
}
TEST(RtcEventLogTest, LogSessionAndReadBackWith2Extensions) {
RtpHeaderExtensionMap extensions;
extensions.Register(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId);
extensions.Register(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
RtcEventLogSessionDescription session(3141592653u /*Random seed*/);
session.GenerateSessionDescription(4, 4, 1, 1, 0, 0, 0, extensions, 0);
session.WriteSession();
session.ReadAndVerifySession();
}
TEST(RtcEventLogTest, LogSessionAndReadBackWithAllExtensions) {
RtpHeaderExtensionMap extensions;
for (uint32_t i = 0; i < kNumExtensions; i++) {
extensions.Register(kExtensionTypes[i], kExtensionIds[i]);
}
RtcEventLogSessionDescription session(2718281828u /*Random seed*/);
session.GenerateSessionDescription(5, 4, 1, 1, 3, 2, 2, extensions, 2);
session.WriteSession();
session.ReadAndVerifySession();
}
TEST(RtcEventLogTest, LogSessionAndReadBackAllCombinations) {
// Try all combinations of header extensions and up to 2 CSRCS.
for (extensions = 0; extensions < (1u << kNumExtensions); extensions++) {
for (uint32_t extension_selection = 0;
extension_selection < (1u << kNumExtensions); extension_selection++) {
RtpHeaderExtensionMap extensions;
for (uint32_t i = 0; i < kNumExtensions; i++) {
if (extension_selection & (1u << i)) {
extensions.Register(kExtensionTypes[i], kExtensionIds[i]);
}
}
for (uint32_t csrcs_count = 0; csrcs_count < 3; csrcs_count++) {
LogSessionAndReadBack(5 + extensions, // Number of RTP packets.
2 + csrcs_count, // Number of RTCP packets.
3 + csrcs_count, // Number of playout events.
1 + csrcs_count, // Number of BWE loss events.
extensions, // Bit vector choosing extensions.
csrcs_count, // Number of contributing sources.
extensions * 3 + csrcs_count + 1); // Random seed.
RtcEventLogSessionDescription session(extension_selection * 3 +
csrcs_count + 1 /*Random seed*/);
session.GenerateSessionDescription(
2 + extension_selection, // Number of incoming RTP packets.
2 + extension_selection, // Number of outgoing RTP packets.
1 + csrcs_count, // Number of incoming RTCP packets.
1 + csrcs_count, // Number of outgoing RTCP packets.
3 + csrcs_count, // Number of playout events.
1 + csrcs_count, // Number of BWE loss events.
2 + csrcs_count, // Number of BWE delay events.
extensions, // Bit vector choosing extensions.
csrcs_count); // Number of contributing sources.
session.WriteSession();
session.ReadAndVerifySession();
}
}
}
@ -436,8 +705,8 @@ TEST(RtcEventLogTest, LogEventAndReadBack) {
// Create one RTP and one RTCP packet containing random data.
size_t packet_size = prng.Rand(1000, 1100);
RtpPacketToSend rtp_packet =
GenerateRtpPacket(nullptr, 0, packet_size, &prng);
RtpPacketReceived rtp_packet =
GenerateIncomingRtpPacket(nullptr, 0, packet_size, &prng);
rtc::Buffer rtcp_packet = GenerateRtcpPacket(&prng);
// Find the name of the current test, in order to use it as a temporary
@ -451,15 +720,13 @@ TEST(RtcEventLogTest, LogEventAndReadBack) {
fake_clock.SetTimeMicros(prng.Rand<uint32_t>());
std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
log_dumper->LogRtpHeader(kIncomingPacket, rtp_packet.data(),
rtp_packet.size());
log_dumper->LogIncomingRtpHeader(rtp_packet);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->StartLogging(temp_filename, 10000000);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->LogRtcpPacket(kOutgoingPacket, rtcp_packet.data(),
rtcp_packet.size());
log_dumper->LogOutgoingRtcpPacket(rtcp_packet);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->StopLogging();
@ -474,9 +741,7 @@ TEST(RtcEventLogTest, LogEventAndReadBack) {
RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
RtcEventLogTestHelper::VerifyRtpEvent(
parsed_log, 1, kIncomingPacket, rtp_packet.data(),
rtp_packet.headers_size(), rtp_packet.size());
RtcEventLogTestHelper::VerifyIncomingRtpEvent(parsed_log, 1, rtp_packet);
RtcEventLogTestHelper::VerifyRtcpEvent(
parsed_log, 2, kOutgoingPacket, rtcp_packet.data(), rtcp_packet.size());
@ -721,7 +986,7 @@ class ConfigReadWriteTest {
public:
ConfigReadWriteTest() : prng(987654321) {}
virtual ~ConfigReadWriteTest() {}
virtual void GenerateConfig(uint32_t extensions_bitvector) = 0;
virtual void GenerateConfig(const RtpHeaderExtensionMap& extensions) = 0;
virtual void VerifyConfig(const ParsedRtcEventLog& parsed_log,
size_t index) = 0;
virtual void LogConfig(RtcEventLog* event_log) = 0;
@ -734,8 +999,11 @@ class ConfigReadWriteTest {
test::OutputPath() + test_info->test_case_name() + test_info->name();
// Use all extensions.
uint32_t extensions_bitvector = (1u << kNumExtensions) - 1;
GenerateConfig(extensions_bitvector);
RtpHeaderExtensionMap extensions;
for (uint32_t i = 0; i < kNumExtensions; i++) {
extensions.Register(kExtensionTypes[i], kExtensionIds[i]);
}
GenerateConfig(extensions);
// Log a single config event and stop logging.
rtc::ScopedFakeClock fake_clock;
@ -768,8 +1036,8 @@ class ConfigReadWriteTest {
class AudioReceiveConfigReadWriteTest : public ConfigReadWriteTest {
public:
void GenerateConfig(uint32_t extensions_bitvector) override {
GenerateAudioReceiveConfig(extensions_bitvector, &config, &prng);
void GenerateConfig(const RtpHeaderExtensionMap& extensions) override {
GenerateAudioReceiveConfig(extensions, &config, &prng);
}
void LogConfig(RtcEventLog* event_log) override {
event_log->LogAudioReceiveStreamConfig(config);
@ -785,8 +1053,8 @@ class AudioReceiveConfigReadWriteTest : public ConfigReadWriteTest {
class AudioSendConfigReadWriteTest : public ConfigReadWriteTest {
public:
AudioSendConfigReadWriteTest() {}
void GenerateConfig(uint32_t extensions_bitvector) override {
GenerateAudioSendConfig(extensions_bitvector, &config, &prng);
void GenerateConfig(const RtpHeaderExtensionMap& extensions) override {
GenerateAudioSendConfig(extensions, &config, &prng);
}
void LogConfig(RtcEventLog* event_log) override {
event_log->LogAudioSendStreamConfig(config);
@ -802,8 +1070,8 @@ class AudioSendConfigReadWriteTest : public ConfigReadWriteTest {
class VideoReceiveConfigReadWriteTest : public ConfigReadWriteTest {
public:
VideoReceiveConfigReadWriteTest() {}
void GenerateConfig(uint32_t extensions_bitvector) override {
GenerateVideoReceiveConfig(extensions_bitvector, &config, &prng);
void GenerateConfig(const RtpHeaderExtensionMap& extensions) override {
GenerateVideoReceiveConfig(extensions, &config, &prng);
}
void LogConfig(RtcEventLog* event_log) override {
event_log->LogVideoReceiveStreamConfig(config);
@ -819,8 +1087,8 @@ class VideoReceiveConfigReadWriteTest : public ConfigReadWriteTest {
class VideoSendConfigReadWriteTest : public ConfigReadWriteTest {
public:
VideoSendConfigReadWriteTest() {}
void GenerateConfig(uint32_t extensions_bitvector) override {
GenerateVideoSendConfig(extensions_bitvector, &config, &prng);
void GenerateConfig(const RtpHeaderExtensionMap& extensions) override {
GenerateVideoSendConfig(extensions, &config, &prng);
}
void LogConfig(RtcEventLog* event_log) override {
event_log->LogVideoSendStreamConfig(config);
@ -835,8 +1103,8 @@ class VideoSendConfigReadWriteTest : public ConfigReadWriteTest {
class AudioNetworkAdaptationReadWriteTest : public ConfigReadWriteTest {
public:
void GenerateConfig(uint32_t extensions_bitvector) override {
GenerateAudioNetworkAdaptation(extensions_bitvector, &config, &prng);
void GenerateConfig(const RtpHeaderExtensionMap& extensions) override {
GenerateAudioNetworkAdaptation(extensions, &config, &prng);
}
void LogConfig(RtcEventLog* event_log) override {
event_log->LogAudioNetworkAdaptation(config);

61
logging/rtc_event_log/rtc_event_log_unittest_helper.cc
View File

@ -18,6 +18,7 @@
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "rtc_base/checks.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@ -349,25 +350,23 @@ void RtcEventLogTestHelper::VerifyAudioSendStreamConfig(
VerifyStreamConfigsAreEqual(config, parsed_config);
}
void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
PacketDirection direction,
const uint8_t* header,
size_t header_size,
size_t total_size) {
void RtcEventLogTestHelper::VerifyIncomingRtpEvent(
const ParsedRtcEventLog& parsed_log,
size_t index,
const RtpPacketReceived& expected_packet) {
const rtclog::Event& event = parsed_log.events_[index];
ASSERT_TRUE(IsValidBasicEvent(event));
ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
ASSERT_TRUE(rtp_packet.has_incoming());
EXPECT_EQ(direction == kIncomingPacket, rtp_packet.incoming());
EXPECT_TRUE(rtp_packet.incoming());
ASSERT_TRUE(rtp_packet.has_packet_length());
EXPECT_EQ(total_size, rtp_packet.packet_length());
EXPECT_EQ(expected_packet.size(), rtp_packet.packet_length());
size_t header_size = expected_packet.headers_size();
ASSERT_TRUE(rtp_packet.has_header());
ASSERT_EQ(header_size, rtp_packet.header().size());
for (size_t i = 0; i < header_size; i++) {
EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i]));
}
EXPECT_THAT(testing::make_tuple(expected_packet.data(), header_size),
testing::ElementsAreArray(rtp_packet.header().data(),
rtp_packet.header().size()));
// Check consistency of the parser.
PacketDirection parsed_direction;
@ -375,10 +374,40 @@ void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log,
size_t 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);
ASSERT_EQ(header_size, parsed_header_size);
EXPECT_EQ(0, std::memcmp(header, parsed_header, header_size));
EXPECT_EQ(total_size, parsed_total_size);
EXPECT_EQ(kIncomingPacket, parsed_direction);
EXPECT_THAT(testing::make_tuple(expected_packet.data(), header_size),
testing::ElementsAreArray(parsed_header, parsed_header_size));
EXPECT_EQ(expected_packet.size(), parsed_total_size);
}
void RtcEventLogTestHelper::VerifyOutgoingRtpEvent(
const ParsedRtcEventLog& parsed_log,
size_t index,
const RtpPacketToSend& expected_packet) {
const rtclog::Event& event = parsed_log.events_[index];
ASSERT_TRUE(IsValidBasicEvent(event));
ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
ASSERT_TRUE(rtp_packet.has_incoming());
EXPECT_FALSE(rtp_packet.incoming());
ASSERT_TRUE(rtp_packet.has_packet_length());
EXPECT_EQ(expected_packet.size(), rtp_packet.packet_length());
size_t header_size = expected_packet.headers_size();
ASSERT_TRUE(rtp_packet.has_header());
EXPECT_THAT(testing::make_tuple(expected_packet.data(), header_size),
testing::ElementsAreArray(rtp_packet.header().data(),
rtp_packet.header().size()));
// Check consistency of the parser.
PacketDirection parsed_direction;
uint8_t parsed_header[1500];
size_t parsed_header_size, parsed_total_size;
parsed_log.GetRtpHeader(index, &parsed_direction, parsed_header,
&parsed_header_size, &parsed_total_size);
EXPECT_EQ(kOutgoingPacket, parsed_direction);
EXPECT_THAT(testing::make_tuple(expected_packet.data(), header_size),
testing::ElementsAreArray(parsed_header, parsed_header_size));
EXPECT_EQ(expected_packet.size(), parsed_total_size);
}
void RtcEventLogTestHelper::VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,

14
logging/rtc_event_log/rtc_event_log_unittest_helper.h
View File

@ -13,6 +13,8 @@
#include "call/call.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
namespace webrtc {
@ -32,12 +34,12 @@ class RtcEventLogTestHelper {
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,
const uint8_t* header,
size_t header_size,
size_t total_size);
static void VerifyIncomingRtpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
const RtpPacketReceived& expected_packet);
static void VerifyOutgoingRtpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
const RtpPacketToSend& expected_packet);
static void VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
size_t index,
PacketDirection direction,