Fix audio/video sync when FEC is enabled.
Also improves the tests by adding a test case for FEC, and running the a/v sync
tests with NACK and simulated packet loss.
BUG=crbug/374104
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index e29b8de..ccad755 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -159,6 +159,8 @@
const uint8_t CallTest::kSendPayloadType = 100;
const uint8_t CallTest::kFakeSendPayloadType = 125;
const uint8_t CallTest::kSendRtxPayloadType = 98;
+const uint8_t CallTest::kRedPayloadType = 118;
+const uint8_t CallTest::kUlpfecPayloadType = 119;
const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
0xBADCAFF};
const uint32_t CallTest::kSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF};
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index b390329..a03c3bb 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -36,6 +36,8 @@
static const uint8_t kSendPayloadType;
static const uint8_t kSendRtxPayloadType;
static const uint8_t kFakeSendPayloadType;
+ static const uint8_t kRedPayloadType;
+ static const uint8_t kUlpfecPayloadType;
static const uint32_t kSendRtxSsrcs[kNumSsrcs];
static const uint32_t kSendSsrcs[kNumSsrcs];
static const uint32_t kReceiverLocalSsrc;
diff --git a/webrtc/video/call_perf_tests.cc b/webrtc/video/call_perf_tests.cc
index 0eaf310..3d1bf7f 100644
--- a/webrtc/video/call_perf_tests.cc
+++ b/webrtc/video/call_perf_tests.cc
@@ -48,6 +48,8 @@
class CallPerfTest : public test::CallTest {
protected:
+ void TestAudioVideoSync(bool fec);
+
void TestMinTransmitBitrate(bool pad_to_min_bitrate);
void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
@@ -188,7 +190,7 @@
int64_t first_time_in_sync_;
};
-TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) {
+void CallPerfTest::TestAudioVideoSync(bool fec) {
class AudioPacketReceiver : public PacketReceiver {
public:
AudioPacketReceiver(int channel, VoENetwork* voe_network)
@@ -229,6 +231,7 @@
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 500;
+ net_config.loss_percent = 5;
SyncRtcpObserver audio_observer(net_config);
VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(),
channel,
@@ -257,6 +260,14 @@
CreateSendConfig(1);
CreateMatchingReceiveConfigs();
+ send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+ if (fec) {
+ send_config_.rtp.fec.red_payload_type = kRedPayloadType;
+ send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+ receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
+ receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+ }
+ receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
receive_configs_[0].renderer = &observer;
receive_configs_[0].audio_channel_id = channel;
@@ -294,6 +305,14 @@
VoiceEngine::Delete(voice_engine);
}
+TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) {
+ TestAudioVideoSync(false);
+}
+
+TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) {
+ TestAudioVideoSync(true);
+}
+
void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
int threshold_ms,
int start_time_ms,
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 11ce642..114ab65 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -44,8 +44,6 @@
namespace webrtc {
static const unsigned long kSilenceTimeoutMs = 2000;
-static const int kRedPayloadType = 118;
-static const int kUlpfecPayloadType = 119;
class EndToEndTest : public test::CallTest {
public:
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 76e9f0f..b628312 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -311,8 +311,6 @@
}
TEST_F(VideoSendStreamTest, SupportsFec) {
- static const int kRedPayloadType = 118;
- static const int kUlpfecPayloadType = 119;
class FecObserver : public test::SendTest {
public:
FecObserver()
@@ -486,8 +484,6 @@
static const uint32_t start = 90;
static const uint32_t stop = 290;
- static const int kRedPayloadType = 118;
- static const int kUlpfecPayloadType = 119;
// Observer that verifies that the expected number of packets and bytes
// arrive for each frame size, from start_size to stop_size.
class FrameFragmentationTest : public test::SendTest,
diff --git a/webrtc/video_engine/vie_receiver.cc b/webrtc/video_engine/vie_receiver.cc
index 5d90ac6..a19c4b0 100644
--- a/webrtc/video_engine/vie_receiver.cc
+++ b/webrtc/video_engine/vie_receiver.cc
@@ -193,7 +193,8 @@
return false;
}
header.payload_type_frequency = kVideoPayloadTypeFrequency;
- return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
+ bool in_order = IsPacketInOrder(header);
+ return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
}
void ViEReceiver::ReceivedBWEPacket(
