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Removed cname and receiver_reference_time_report from proto and logging code. Changed logging of RTCP to omit messages of type SDES and APP.

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BUG=

Review URL: https://codereview.webrtc.org/1419523004

Cr-Commit-Position: refs/heads/master@{#10542}
This commit is contained in:
terelius
2015-11-06 09:00:18 -08:00
committed by Commit bot
parent 56b1128c8f
commit d66daa2d2f
3 changed files with 86 additions and 56 deletions
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54
webrtc/call/rtc_event_log.cc
View File

@ -17,7 +17,9 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
@ -269,9 +271,6 @@ void RtcEventLogImpl::LogVideoReceiveStreamConfig(
receiver_config->set_local_ssrc(config.rtp.local_ssrc);
receiver_config->set_rtcp_mode(ConvertRtcpMode(config.rtp.rtcp_mode));
receiver_config->set_receiver_reference_time_report(
config.rtp.rtcp_xr.receiver_reference_time_report);
receiver_config->set_remb(config.rtp.remb);
for (const auto& kv : config.rtp.rtx) {
@ -322,8 +321,6 @@ void RtcEventLogImpl::LogVideoSendStreamConfig(
}
sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type);
sender_config->set_c_name(config.rtp.c_name);
rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
encoder->set_name(config.encoder_settings.payload_name);
encoder->set_payload_type(config.encoder_settings.payload_type);
@ -371,7 +368,52 @@ void RtcEventLogImpl::LogRtcpPacket(bool incoming,
rtcp_event.set_type(rtclog::Event::RTCP_EVENT);
rtcp_event.mutable_rtcp_packet()->set_incoming(incoming);
rtcp_event.mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
rtcp_event.mutable_rtcp_packet()->set_packet_data(packet, length);
RTCPUtility::RtcpCommonHeader header;
const uint8_t* block_begin = packet;
const uint8_t* packet_end = packet + length;
RTC_DCHECK(length <= IP_PACKET_SIZE);
uint8_t buffer[IP_PACKET_SIZE];
uint32_t buffer_length = 0;
while (block_begin < packet_end) {
if (!RtcpParseCommonHeader(block_begin, packet_end - block_begin,
&header)) {
break; // Incorrect message header.
}
uint32_t block_size = header.BlockSize();
switch (header.packet_type) {
case RTCPUtility::PT_SR:
FALLTHROUGH();
case RTCPUtility::PT_RR:
FALLTHROUGH();
case RTCPUtility::PT_BYE:
FALLTHROUGH();
case RTCPUtility::PT_IJ:
FALLTHROUGH();
case RTCPUtility::PT_RTPFB:
FALLTHROUGH();
case RTCPUtility::PT_PSFB:
FALLTHROUGH();
case RTCPUtility::PT_XR:
// We log sender reports, receiver reports, bye messages
// inter-arrival jitter, third-party loss reports, payload-specific
// feedback and extended reports.
memcpy(buffer + buffer_length, block_begin, block_size);
buffer_length += block_size;
break;
case RTCPUtility::PT_SDES:
FALLTHROUGH();
case RTCPUtility::PT_APP:
FALLTHROUGH();
default:
// We don't log sender descriptions, application defined messages
// or message blocks of unknown type.
break;
}
block_begin += block_size;
}
rtcp_event.mutable_rtcp_packet()->set_packet_data(buffer, buffer_length);
HandleEvent(&rtcp_event);
}

16
webrtc/call/rtc_event_log.proto
View File

@ -137,20 +137,17 @@ message VideoReceiveConfig {
// required - RTCP mode to use.
optional RtcpMode rtcp_mode = 3;
// required - Extended RTCP settings.
optional bool receiver_reference_time_report = 4;
// required - Receiver estimated maximum bandwidth.
optional bool remb = 5;
optional bool remb = 4;
// Map from video RTP payload type -> RTX config.
repeated RtxMap rtx_map = 6;
repeated RtxMap rtx_map = 5;
// RTP header extensions used for the received stream.
repeated RtpHeaderExtension header_extensions = 7;
repeated RtpHeaderExtension header_extensions = 6;
// List of decoders associated with the stream.
repeated DecoderConfig decoders = 8;
repeated DecoderConfig decoders = 7;
}
@ -209,11 +206,8 @@ message VideoSendConfig {
// required if rtx_ssrcs is used - Payload type for retransmitted packets.
optional int32 rtx_payload_type = 4;
// required - Canonical end-point identifier.
optional string c_name = 5;
// required - Encoder associated with the stream.
optional EncoderConfig encoder = 6;
optional EncoderConfig encoder = 5;
}

72
webrtc/call/rtc_event_log_unittest.cc
View File

@ -20,6 +20,7 @@
#include "webrtc/base/thread.h"
#include "webrtc/call.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/random.h"
@ -138,9 +139,6 @@ void VerifyReceiveStreamConfig(const rtclog::Event& event,
else
EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
receiver_config.rtcp_mode());
ASSERT_TRUE(receiver_config.has_receiver_reference_time_report());
EXPECT_EQ(config.rtp.rtcp_xr.receiver_reference_time_report,
receiver_config.receiver_reference_time_report());
ASSERT_TRUE(receiver_config.has_remb());
EXPECT_EQ(config.rtp.remb, receiver_config.remb());
// Check RTX map.
@ -214,9 +212,6 @@ void VerifySendStreamConfig(const rtclog::Event& event,
ASSERT_TRUE(sender_config.has_rtx_payload_type());
EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
}
// Check CNAME.
ASSERT_TRUE(sender_config.has_c_name());
EXPECT_EQ(config.rtp.c_name, sender_config.c_name());
// Check encoder.
ASSERT_TRUE(sender_config.has_encoder());
ASSERT_TRUE(sender_config.encoder().has_name());
@ -230,7 +225,7 @@ void VerifySendStreamConfig(const rtclog::Event& event,
void VerifyRtpEvent(const rtclog::Event& event,
bool incoming,
MediaType media_type,
uint8_t* header,
const uint8_t* header,
size_t header_size,
size_t total_size) {
ASSERT_TRUE(IsValidBasicEvent(event));
@ -252,7 +247,7 @@ void VerifyRtpEvent(const rtclog::Event& event,
void VerifyRtcpEvent(const rtclog::Event& event,
bool incoming,
MediaType media_type,
uint8_t* packet,
const uint8_t* packet,
size_t total_size) {
ASSERT_TRUE(IsValidBasicEvent(event));
ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
@ -353,12 +348,19 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
return header_size;
}
void GenerateRtcpPacket(uint8_t* packet,
size_t packet_size,
test::Random* prng) {
for (size_t i = 0; i < packet_size; i++) {
packet[i] = prng->Rand<uint8_t>();
}
rtc::scoped_ptr<rtcp::RawPacket> GenerateRtcpPacket(test::Random* prng) {
rtcp::ReportBlock report_block;
report_block.To(prng->Rand<uint32_t>()); // Remote SSRC.
report_block.WithFractionLost(prng->Rand(50));
rtcp::SenderReport sender_report;
sender_report.From(prng->Rand<uint32_t>()); // Sender SSRC.
sender_report.WithNtpSec(prng->Rand<uint32_t>());
sender_report.WithNtpFrac(prng->Rand<uint32_t>());
sender_report.WithPacketCount(prng->Rand<uint32_t>());
sender_report.WithReportBlock(report_block);
return sender_report.Build();
}
void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
@ -375,7 +377,6 @@ void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
// Add extensions and settings for RTCP.
config->rtp.rtcp_mode =
prng->Rand<bool>() ? RtcpMode::kCompound : RtcpMode::kReducedSize;
config->rtp.rtcp_xr.receiver_reference_time_report = prng->Rand<bool>();
config->rtp.remb = prng->Rand<bool>();
// Add a map from a payload type to a new ssrc and a new payload type for RTX.
VideoReceiveStream::Config::Rtp::Rtx rtx_pair;
@ -402,8 +403,6 @@ void GenerateVideoSendConfig(uint32_t extensions_bitvector,
// Add a map from a payload type to new ssrcs and a new payload type for RTX.
config->rtp.rtx.ssrcs.push_back(prng->Rand<uint32_t>());
config->rtp.rtx.payload_type = prng->Rand(0, 127);
// Add a CNAME.
config->rtp.c_name = "some.user@some.host";
// Add header extensions.
for (unsigned i = 0; i < kNumExtensions; i++) {
if (extensions_bitvector & (1u << i)) {
@ -426,7 +425,7 @@ void LogSessionAndReadBack(size_t rtp_count,
ASSERT_LE(playout_count, rtp_count);
ASSERT_LE(bwe_loss_count, rtp_count);
std::vector<rtc::Buffer> rtp_packets;
std::vector<rtc::Buffer> rtcp_packets;
std::vector<rtc::scoped_ptr<rtcp::RawPacket> > rtcp_packets;
std::vector<size_t> rtp_header_sizes;
std::vector<uint32_t> playout_ssrcs;
std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates;
@ -447,9 +446,7 @@ void LogSessionAndReadBack(size_t rtp_count,
}
// Create rtcp_count RTCP packets containing random data.
for (size_t i = 0; i < rtcp_count; i++) {
size_t packet_size = prng.Rand(1000, 1100);
rtcp_packets.push_back(rtc::Buffer(packet_size));
GenerateRtcpPacket(rtcp_packets[i].data(), packet_size, &prng);
rtcp_packets.push_back(GenerateRtcpPacket(&prng));
}
// Create playout_count random SSRCs to use when logging AudioPlayout events.
for (size_t i = 0; i < playout_count; i++) {
@ -457,7 +454,8 @@ void LogSessionAndReadBack(size_t rtp_count,
}
// Create bwe_loss_count random bitrate updates for BwePacketLoss.
for (size_t i = 0; i < bwe_loss_count; i++) {
bwe_loss_updates.push_back(std::pair<int32_t, uint8_t>(rand(), rand()));
bwe_loss_updates.push_back(
std::make_pair(prng.Rand<int32_t>(), prng.Rand<uint8_t>()));
}
// Create configurations for the video streams.
GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
@ -488,8 +486,8 @@ void LogSessionAndReadBack(size_t rtp_count,
log_dumper->LogRtcpPacket(
rtcp_index % 2 == 0, // Every second packet is incoming
rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
rtcp_packets[rtcp_index - 1].data(),
rtcp_packets[rtcp_index - 1].size());
rtcp_packets[rtcp_index - 1]->Buffer(),
rtcp_packets[rtcp_index - 1]->Length());
rtcp_index++;
}
if (i * playout_count >= playout_index * rtp_count) {
@ -536,8 +534,8 @@ void LogSessionAndReadBack(size_t rtp_count,
VerifyRtcpEvent(parsed_stream.stream(event_index),
rtcp_index % 2 == 0, // Every second packet is incoming.
rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
rtcp_packets[rtcp_index - 1].data(),
rtcp_packets[rtcp_index - 1].size());
rtcp_packets[rtcp_index - 1]->Buffer(),
rtcp_packets[rtcp_index - 1]->Length());
event_index++;
rtcp_index++;
}
@ -604,8 +602,8 @@ void DropOldEvents(uint32_t extensions_bitvector,
unsigned int random_seed) {
rtc::Buffer old_rtp_packet;
rtc::Buffer recent_rtp_packet;
rtc::Buffer old_rtcp_packet;
rtc::Buffer recent_rtcp_packet;
rtc::scoped_ptr<rtcp::RawPacket> old_rtcp_packet;
rtc::scoped_ptr<rtcp::RawPacket> recent_rtcp_packet;
VideoReceiveStream::Config receiver_config(nullptr);
VideoSendStream::Config sender_config(nullptr);
@ -624,12 +622,8 @@ void DropOldEvents(uint32_t extensions_bitvector,
recent_rtp_packet.data(), packet_size, &prng);
// Create two RTCP packets containing random data.
packet_size = prng.Rand(1000, 1100);
old_rtcp_packet.SetSize(packet_size);
GenerateRtcpPacket(old_rtcp_packet.data(), packet_size, &prng);
packet_size = prng.Rand(1000, 1100);
recent_rtcp_packet.SetSize(packet_size);
GenerateRtcpPacket(recent_rtcp_packet.data(), packet_size, &prng);
old_rtcp_packet = GenerateRtcpPacket(&prng);
recent_rtcp_packet = GenerateRtcpPacket(&prng);
// Create configurations for the video streams.
GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
@ -650,16 +644,16 @@ void DropOldEvents(uint32_t extensions_bitvector,
log_dumper->LogVideoSendStreamConfig(sender_config);
log_dumper->LogRtpHeader(false, MediaType::AUDIO, old_rtp_packet.data(),
old_rtp_packet.size());
log_dumper->LogRtcpPacket(true, MediaType::AUDIO, old_rtcp_packet.data(),
old_rtcp_packet.size());
log_dumper->LogRtcpPacket(true, MediaType::AUDIO, old_rtcp_packet->Buffer(),
old_rtcp_packet->Length());
// Sleep 55 ms to let old events be removed from the queue.
rtc::Thread::SleepMs(55);
log_dumper->StartLogging(temp_filename, 10000000);
log_dumper->LogRtpHeader(true, MediaType::VIDEO, recent_rtp_packet.data(),
recent_rtp_packet.size());
log_dumper->LogRtcpPacket(false, MediaType::VIDEO,
recent_rtcp_packet.data(),
recent_rtcp_packet.size());
recent_rtcp_packet->Buffer(),
recent_rtcp_packet->Length());
}
// Read the generated file from disk.
@ -677,7 +671,7 @@ void DropOldEvents(uint32_t extensions_bitvector,
recent_rtp_packet.data(), recent_header_size,
recent_rtp_packet.size());
VerifyRtcpEvent(parsed_stream.stream(4), false, MediaType::VIDEO,
recent_rtcp_packet.data(), recent_rtcp_packet.size());
recent_rtcp_packet->Buffer(), recent_rtcp_packet->Length());
// Clean up temporary file - can be pretty slow.
remove(temp_filename.c_str());