namespace webrtc {
namespace {
- static const int kAbsoluteSendTimeExtensionId = 7;
- static const int kMaxPacketSize = 1500;
+static const int kAbsoluteSendTimeExtensionId = 7;
+static const int kMaxPacketSize = 1500;
}
class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
rtp_parser_(RtpHeaderParser::Create()),
feedback_transport_(feedback_transport),
receive_stats_(ReceiveStatistics::Create(clock)),
- payload_registry_(new RTPPayloadRegistry(
- -1, RTPPayloadStrategy::CreateStrategy(false))),
+ payload_registry_(
+ new RTPPayloadRegistry(-1,
+ RTPPayloadStrategy::CreateStrategy(false))),
clock_(clock),
num_expected_ssrcs_(num_expected_ssrcs),
rtx_media_ssrcs_(rtx_media_ssrcs),
// be able to produce an RTCP with REMB.
RtpRtcp::Configuration config;
config.receive_statistics = receive_stats_.get();
+ feedback_transport_.Enable();
config.outgoing_transport = &feedback_transport_;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetREMBStatus(true);
if (ssrcs.size() == num_expected_ssrcs_ && bitrate >= kExpectedBitrateBps) {
if (rtx_media_ssrcs_.empty() || rtx_media_sent_ > 0) {
const ::testing::TestInfo* const test_info =
- ::testing::UnitTest::GetInstance()->current_test_info();
- webrtc::test::PrintResult("total-sent", "", test_info->name(),
- total_sent_, "bytes", false);
- webrtc::test::PrintResult("padding-sent", "", test_info->name(),
- padding_sent_, "bytes", false);
- webrtc::test::PrintResult("rtx-media-sent", "", test_info->name(),
- rtx_media_sent_, "bytes", false);
- webrtc::test::PrintResult("total-packets-sent", "", test_info->name(),
- total_packets_sent_, "packets", false);
- webrtc::test::PrintResult("padding-packets-sent", "", test_info->name(),
- padding_packets_sent_, "packets", false);
- webrtc::test::PrintResult("rtx-packets-sent", "", test_info->name(),
- rtx_media_packets_sent_, "packets", false);
+ ::testing::UnitTest::GetInstance()->current_test_info();
+ webrtc::test::PrintResult(
+ "total-sent", "", test_info->name(), total_sent_, "bytes", false);
+ webrtc::test::PrintResult("padding-sent",
+ "",
+ test_info->name(),
+ padding_sent_,
+ "bytes",
+ false);
+ webrtc::test::PrintResult("rtx-media-sent",
+ "",
+ test_info->name(),
+ rtx_media_sent_,
+ "bytes",
+ false);
+ webrtc::test::PrintResult("total-packets-sent",
+ "",
+ test_info->name(),
+ total_packets_sent_,
+ "packets",
+ false);
+ webrtc::test::PrintResult("padding-packets-sent",
+ "",
+ test_info->name(),
+ padding_packets_sent_,
+ "packets",
+ false);
+ webrtc::test::PrintResult("rtx-packets-sent",
+ "",
+ test_info->name(),
+ rtx_media_packets_sent_,
+ "packets",
+ false);
all_ssrcs_sent_->Set();
}
}
uint8_t restored_packet[kMaxPacketSize];
uint8_t* restored_packet_ptr = restored_packet;
int restored_length = static_cast<int>(length);
- payload_registry_->RestoreOriginalPacket(
- &restored_packet_ptr, packet, &restored_length,
- rtx_media_ssrcs_[header.ssrc],
- header);
+ payload_registry_->RestoreOriginalPacket(&restored_packet_ptr,
+ packet,
+ &restored_length,
+ rtx_media_ssrcs_[header.ssrc],
+ header);
length = restored_length;
- EXPECT_TRUE(rtp_parser_->Parse(restored_packet, static_cast<int>(length),
- &header));
+ EXPECT_TRUE(rtp_parser_->Parse(
+ restored_packet, static_cast<int>(length), &header));
} else {
rtp_rtcp_->SetRemoteSSRC(header.ssrc);
}
}
test::DirectTransport receiver_transport;
int num_expected_ssrcs = kNumberOfStreams + (rtx ? 1 : 0);
- StreamObserver stream_observer(
- num_expected_ssrcs, rtx_ssrc_map, &receiver_transport,
- Clock::GetRealTimeClock());
+ StreamObserver stream_observer(num_expected_ssrcs,
+ rtx_ssrc_map,
+ &receiver_transport,
+ Clock::GetRealTimeClock());
Call::Config call_config(&stream_observer);
webrtc::Config webrtc_config;
send_config.codec.plType = 125;
send_config.pacing = pacing;
send_config.rtp.nack.rtp_history_ms = 1000;
- send_config.rtp.ssrcs.insert(send_config.rtp.ssrcs.begin(), ssrcs.begin(),
- ssrcs.end());
+ send_config.rtp.ssrcs.insert(
+ send_config.rtp.ssrcs.begin(), ssrcs.begin(), ssrcs.end());
if (rtx) {
- send_config.rtp.rtx.rtx_payload_type = 96;
+ send_config.rtp.rtx.payload_type = 96;
send_config.rtp.rtx.ssrcs.insert(send_config.rtp.rtx.ssrcs.begin(),
kRtxSsrcs,
kRtxSsrcs + kNumberOfStreams);
std::map<uint32_t, bool> reserved_ssrcs_;
};
-TEST_F(RampUpTest, WithoutPacing) {
- RunRampUpTest(false, false);
-}
+TEST_F(RampUpTest, WithoutPacing) { RunRampUpTest(false, false); }
-TEST_F(RampUpTest, WithPacing) {
- RunRampUpTest(true, false);
-}
+TEST_F(RampUpTest, WithPacing) { RunRampUpTest(true, false); }
-TEST_F(RampUpTest, WithPacingAndRtx) {
- RunRampUpTest(true, true);
-}
+TEST_F(RampUpTest, WithPacingAndRtx) { RunRampUpTest(true, true); }
} // namespace webrtc