webrtc/call/rtc_event_log_unittest.cc - Issue 1419523004: Skip logging RTCP messages of type SDES and APP.

Unified Diff: webrtc/call/rtc_event_log_unittest.cc

Issue 1419523004: Skip logging RTCP messages of type SDES and APP. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Major rebase Created 5 years, 1 month ago

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Index: webrtc/call/rtc_event_log_unittest.cc

diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc

index 973e5424eb0ea388eca8c6da6463026b92415879..9dc7bec215ad26a33a5ec623bc2c4a72cbb92509 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 @@ 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());

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