Removed cname and receiver_reference_time_report from proto and logging code. Changed logging of RTCP to omit messages of type SDES and APP.
BUG=
Review URL: https://codereview.webrtc.org/1419523004
Cr-Commit-Position: refs/heads/master@{#10542}
diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc
index abc2eb4..9f592ce 100644
--- a/webrtc/call/rtc_event_log.cc
+++ b/webrtc/call/rtc_event_log.cc
@@ -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 @@
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 @@
}
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 @@
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);
}
diff --git a/webrtc/call/rtc_event_log.proto b/webrtc/call/rtc_event_log.proto
index e8ec99d..b14306e 100644
--- a/webrtc/call/rtc_event_log.proto
+++ b/webrtc/call/rtc_event_log.proto
@@ -137,20 +137,17 @@
// 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 @@
// 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;
}
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
index 973e542..9dc7bec 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -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 @@
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 @@
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 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 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 @@
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 @@
// 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 @@
// 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 @@
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 @@
}
// 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 @@
}
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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());
