Add field trial to force playout delay
This CL adds the field trial WebRTC-ForcePlayoutDelay with parameters
min_ms and max_ms. If both of these values are set, the playout delay
of any received packet will be overridden by the specified values.
Bug: None
Change-Id: I353282097e3ffa437dfc5affdfdf7780b09474e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174180
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31149}
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 1ae5c5e..e1dd736 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -237,6 +237,8 @@
process_thread_(process_thread),
ntp_estimator_(clock),
rtp_header_extensions_(config_.rtp.extensions),
+ forced_playout_delay_max_ms_("max_ms", absl::nullopt),
+ forced_playout_delay_min_ms_("min_ms", absl::nullopt),
rtp_receive_statistics_(rtp_receive_statistics),
ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc,
this,
@@ -290,6 +292,10 @@
if (config_.rtp.rtcp_xr.receiver_reference_time_report)
rtp_rtcp_->SetRtcpXrRrtrStatus(true);
+ ParseFieldTrial(
+ {&forced_playout_delay_max_ms_, &forced_playout_delay_min_ms_},
+ field_trial::FindFullName("WebRTC-ForcePlayoutDelay"));
+
process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
if (config_.rtp.lntf.enabled) {
@@ -513,7 +519,12 @@
rtp_packet.GetExtension<VideoContentTypeExtension>(
&video_header.content_type);
rtp_packet.GetExtension<VideoTimingExtension>(&video_header.video_timing);
- rtp_packet.GetExtension<PlayoutDelayLimits>(&video_header.playout_delay);
+ if (forced_playout_delay_max_ms_ && forced_playout_delay_min_ms_) {
+ video_header.playout_delay.max_ms = *forced_playout_delay_max_ms_;
+ video_header.playout_delay.min_ms = *forced_playout_delay_min_ms_;
+ } else {
+ rtp_packet.GetExtension<PlayoutDelayLimits>(&video_header.playout_delay);
+ }
rtp_packet.GetExtension<FrameMarkingExtension>(&video_header.frame_marking);
ParseGenericDependenciesResult generic_descriptor_state =
diff --git a/video/rtp_video_stream_receiver.h b/video/rtp_video_stream_receiver.h
index 3e07df9..0289f23 100644
--- a/video/rtp_video_stream_receiver.h
+++ b/video/rtp_video_stream_receiver.h
@@ -43,6 +43,7 @@
#include "modules/video_coding/unique_timestamp_counter.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
+#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/synchronization/sequence_checker.h"
#include "rtc_base/thread_annotations.h"
@@ -299,6 +300,10 @@
RemoteNtpTimeEstimator ntp_estimator_;
RtpHeaderExtensionMap rtp_header_extensions_;
+ // Set by the field trial WebRTC-ForcePlayoutDelay to override any playout
+ // delay that is specified in the received packets.
+ FieldTrialOptional<int> forced_playout_delay_max_ms_;
+ FieldTrialOptional<int> forced_playout_delay_min_ms_;
ReceiveStatistics* const rtp_receive_statistics_;
std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
diff --git a/video/rtp_video_stream_receiver_unittest.cc b/video/rtp_video_stream_receiver_unittest.cc
index 40602f7..40d63ae 100644
--- a/video/rtp_video_stream_receiver_unittest.cc
+++ b/video/rtp_video_stream_receiver_unittest.cc
@@ -61,6 +61,15 @@
return result;
}
+RTPVideoHeader GetGenericVideoHeader(VideoFrameType frame_type) {
+ RTPVideoHeader video_header;
+ video_header.is_first_packet_in_frame = true;
+ video_header.is_last_packet_in_frame = true;
+ video_header.codec = kVideoCodecGeneric;
+ video_header.frame_type = frame_type;
+ return video_header;
+}
+
class MockTransport : public Transport {
public:
MOCK_METHOD3(SendRtp,
@@ -358,14 +367,11 @@
TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrame) {
RtpPacketReceived rtp_packet;
- RTPVideoHeader video_header;
rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
rtp_packet.SetPayloadType(kPayloadType);
rtp_packet.SetSequenceNumber(1);
- video_header.is_first_packet_in_frame = true;
- video_header.is_last_packet_in_frame = true;
- video_header.codec = kVideoCodecGeneric;
- video_header.frame_type = VideoFrameType::kVideoFrameKey;
+ RTPVideoHeader video_header =
+ GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
data.size());
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
@@ -381,7 +387,6 @@
extension_map.Register<AbsoluteCaptureTimeExtension>(kId0);
RtpPacketReceived rtp_packet(&extension_map);
rtp_packet.SetPayloadType(kPayloadType);
- RTPVideoHeader video_header;
rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
rtp_packet.SetSequenceNumber(1);
rtp_packet.SetTimestamp(1);
@@ -390,10 +395,8 @@
AbsoluteCaptureTime{kAbsoluteCaptureTimestamp,
/*estimated_capture_clock_offset=*/absl::nullopt});
- video_header.is_first_packet_in_frame = true;
- video_header.is_last_packet_in_frame = true;
- video_header.codec = kVideoCodecGeneric;
- video_header.frame_type = VideoFrameType::kVideoFrameKey;
+ RTPVideoHeader video_header =
+ GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
data.size());
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_))
@@ -416,7 +419,6 @@
RtpPacketReceived rtp_packet(&extension_map);
rtp_packet.SetPayloadType(kPayloadType);
- RTPVideoHeader video_header;
rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
uint16_t sequence_number = 1;
uint32_t rtp_timestamp = 1;
@@ -427,10 +429,8 @@
AbsoluteCaptureTime{kAbsoluteCaptureTimestamp,
/*estimated_capture_clock_offset=*/absl::nullopt});
- video_header.is_first_packet_in_frame = true;
- video_header.is_last_packet_in_frame = true;
- video_header.codec = kVideoCodecGeneric;
- video_header.frame_type = VideoFrameType::kVideoFrameKey;
+ RTPVideoHeader video_header =
+ GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
data.size());
EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
@@ -496,13 +496,10 @@
TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrameBitstreamError) {
RtpPacketReceived rtp_packet;
rtp_packet.SetPayloadType(kPayloadType);
- RTPVideoHeader video_header;
rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
rtp_packet.SetSequenceNumber(1);
- video_header.is_first_packet_in_frame = true;
- video_header.is_last_packet_in_frame = true;
- video_header.codec = kVideoCodecGeneric;
- video_header.frame_type = VideoFrameType::kVideoFrameKey;
+ RTPVideoHeader video_header =
+ GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
constexpr uint8_t expected_bitsteam[] = {1, 2, 3, 0xff};
mock_on_complete_frame_callback_.AppendExpectedBitstream(
expected_bitsteam, sizeof(expected_bitsteam));
@@ -658,13 +655,10 @@
TEST_F(RtpVideoStreamReceiverTest, RequestKeyframeIfFirstFrameIsDelta) {
RtpPacketReceived rtp_packet;
rtp_packet.SetPayloadType(kPayloadType);
- RTPVideoHeader video_header;
rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
rtp_packet.SetSequenceNumber(1);
- video_header.is_first_packet_in_frame = true;
- video_header.is_last_packet_in_frame = true;
- video_header.codec = kVideoCodecGeneric;
- video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+ RTPVideoHeader video_header =
+ GetGenericVideoHeader(VideoFrameType::kVideoFrameDelta);
EXPECT_CALL(mock_key_frame_request_sender_, RequestKeyFrame());
rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
video_header);
@@ -675,13 +669,11 @@
RtpPacketReceived rtp_packet;
rtp_packet.SetPayloadType(kPayloadType);
- RTPVideoHeader video_header;
rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
- video_header.is_first_packet_in_frame = true;
+ RTPVideoHeader video_header =
+ GetGenericVideoHeader(VideoFrameType::kVideoFrameDelta);
// Incomplete frames so that the packet buffer is filling up.
video_header.is_last_packet_in_frame = false;
- video_header.codec = kVideoCodecGeneric;
- video_header.frame_type = VideoFrameType::kVideoFrameDelta;
uint16_t start_sequence_number = 1234;
rtp_packet.SetSequenceNumber(start_sequence_number);
while (rtp_packet.SequenceNumber() - start_sequence_number <
@@ -1149,13 +1141,10 @@
RtpPacketReceived rtp_packet;
rtp_packet.SetPayloadType(kPayloadType);
- RTPVideoHeader video_header;
rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
rtp_packet.SetSequenceNumber(1);
- video_header.is_first_packet_in_frame = true;
- video_header.is_last_packet_in_frame = true;
- video_header.codec = kVideoCodecGeneric;
- video_header.frame_type = VideoFrameType::kVideoFrameKey;
+ RTPVideoHeader video_header =
+ GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
data.size());
EXPECT_CALL(*mock_frame_transformer,
@@ -1168,4 +1157,61 @@
receiver = nullptr;
}
+// Test default behavior and when playout delay is overridden by field trial.
+const PlayoutDelay kTransmittedPlayoutDelay = {100, 200};
+const PlayoutDelay kForcedPlayoutDelay = {70, 90};
+struct PlayoutDelayOptions {
+ std::string field_trial;
+ PlayoutDelay expected_delay;
+};
+const PlayoutDelayOptions kDefaultBehavior = {
+ /*field_trial=*/"", /*expected_delay=*/kTransmittedPlayoutDelay};
+const PlayoutDelayOptions kOverridePlayoutDelay = {
+ /*field_trial=*/"WebRTC-ForcePlayoutDelay/min_ms:70,max_ms:90/",
+ /*expected_delay=*/kForcedPlayoutDelay};
+
+class RtpVideoStreamReceiverTestPlayoutDelay
+ : public RtpVideoStreamReceiverTest,
+ public ::testing::WithParamInterface<PlayoutDelayOptions> {
+ protected:
+ RtpVideoStreamReceiverTestPlayoutDelay()
+ : RtpVideoStreamReceiverTest(GetParam().field_trial) {}
+};
+
+INSTANTIATE_TEST_SUITE_P(PlayoutDelay,
+ RtpVideoStreamReceiverTestPlayoutDelay,
+ Values(kDefaultBehavior, kOverridePlayoutDelay));
+
+TEST_P(RtpVideoStreamReceiverTestPlayoutDelay, PlayoutDelay) {
+ rtc::CopyOnWriteBuffer payload_data({1, 2, 3, 4});
+ RtpHeaderExtensionMap extension_map;
+ extension_map.Register<PlayoutDelayLimits>(1);
+ RtpPacketToSend packet_to_send(&extension_map);
+ packet_to_send.SetPayloadType(kPayloadType);
+ packet_to_send.SetSequenceNumber(1);
+
+ // Set playout delay on outgoing packet.
+ EXPECT_TRUE(packet_to_send.SetExtension<PlayoutDelayLimits>(
+ kTransmittedPlayoutDelay));
+ uint8_t* payload = packet_to_send.AllocatePayload(payload_data.size());
+ memcpy(payload, payload_data.data(), payload_data.size());
+
+ RtpPacketReceived received_packet(&extension_map);
+ received_packet.Parse(packet_to_send.data(), packet_to_send.size());
+
+ RTPVideoHeader video_header =
+ GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
+ mock_on_complete_frame_callback_.AppendExpectedBitstream(payload_data.data(),
+ payload_data.size());
+ // Expect the playout delay of encoded frame to be the same as the transmitted
+ // playout delay unless it was overridden by a field trial.
+ EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_))
+ .WillOnce(Invoke([expected_playout_delay = GetParam().expected_delay](
+ video_coding::EncodedFrame* frame) {
+ EXPECT_EQ(frame->EncodedImage().playout_delay_, expected_playout_delay);
+ }));
+ rtp_video_stream_receiver_->OnReceivedPayloadData(
+ received_packet.PayloadBuffer(), received_packet, video_header);
+}
+
} // namespace webrtc
