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Unified Diff: webrtc/video/end_to_end_tests.cc

Issue 2830793005: Reuse allocated encoders in SimulcastEncoderAdapter. (Closed)
Patch Set: Disable new test on tsan due to pre-existing race in libvpx. Created 3 years, 7 months ago
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Index: webrtc/video/end_to_end_tests.cc
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 99e52794240f475ca131c42e70abdc9288529730..e1465f85b5095c39ba2c4cb8b0722d3f37e946c9 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -26,14 +26,19 @@
#include "webrtc/common_video/include/frame_callback.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakevideorenderer.h"
+#include "webrtc/media/base/mediaconstants.h"
+#include "webrtc/media/engine/internalencoderfactory.h"
+#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
+#include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
#include "webrtc/modules/video_coding/include/video_coding_defines.h"
#include "webrtc/system_wrappers/include/metrics.h"
@@ -47,8 +52,8 @@
#include "webrtc/test/field_trial.h"
#include "webrtc/test/frame_generator.h"
#include "webrtc/test/frame_generator_capturer.h"
-#include "webrtc/test/gtest.h"
#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
#include "webrtc/test/null_transport.h"
#include "webrtc/test/rtcp_packet_parser.h"
#include "webrtc/test/rtp_rtcp_observer.h"
@@ -132,6 +137,7 @@ class EndToEndTest : public test::CallTest {
void RespectsRtcpMode(RtcpMode rtcp_mode);
void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp);
+ void TestPictureIdStatePreservation(VideoEncoder* encoder);
void VerifyHistogramStats(bool use_rtx, bool use_red, bool screenshare);
void VerifyNewVideoSendStreamsRespectNetworkState(
MediaType network_to_bring_up,
@@ -3635,8 +3641,8 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
void EndToEndTest::TestRtpStatePreservation(bool use_rtx,
bool provoke_rtcpsr_before_rtp) {
- // This test use other VideoStream settings than the the default settings
- // implemented in DefaultVideoStreamFactory. Therefore this test implement
+ // This test uses other VideoStream settings than the the default settings
+ // implemented in DefaultVideoStreamFactory. Therefore this test implements
// its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
// in ModifyVideoConfigs.
class VideoStreamFactory
@@ -3886,6 +3892,250 @@ TEST_F(EndToEndTest, RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced) {
TestRtpStatePreservation(true, true);
}
+void EndToEndTest::TestPictureIdStatePreservation(VideoEncoder* encoder) {
+ const size_t kFrameMaxWidth = 1280;
+ const size_t kFrameMaxHeight = 720;
+ const size_t kFrameRate = 30;
+
+ // Use a special stream factory in this test to ensure that all simulcast
+ // streams are being sent.
+ class VideoStreamFactory
+ : public VideoEncoderConfig::VideoStreamFactoryInterface {
+ public:
+ VideoStreamFactory() = default;
+
+ private:
+ std::vector<VideoStream> CreateEncoderStreams(
+ int width,
+ int height,
+ const VideoEncoderConfig& encoder_config) override {
+ std::vector<VideoStream> streams =
+ test::CreateVideoStreams(width, height, encoder_config);
+
+ const size_t kBitrate = 100000;
+
+ if (encoder_config.number_of_streams > 1) {
+ RTC_DCHECK_EQ(3, encoder_config.number_of_streams);
+
+ for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
+ streams[i].min_bitrate_bps = kBitrate;
+ streams[i].target_bitrate_bps = kBitrate;
+ streams[i].max_bitrate_bps = kBitrate;
+ }
+
+ // test::CreateVideoStreams does not return frame sizes for the lower
+ // streams that are accepted by VP8Impl::InitEncode.
+ // TODO(brandtr): Fix the problem in test::CreateVideoStreams, rather
+ // than overriding the values here.
+ streams[1].width = streams[2].width / 2;
+ streams[1].height = streams[2].height / 2;
+ streams[0].width = streams[1].width / 2;
+ streams[0].height = streams[1].height / 2;
+ } else {
+ // Use the same total bitrates when sending a single stream to avoid
+ // lowering the bitrate estimate and requiring a subsequent rampup.
+ streams[0].min_bitrate_bps = 3 * kBitrate;
+ streams[0].target_bitrate_bps = 3 * kBitrate;
+ streams[0].max_bitrate_bps = 3 * kBitrate;
+ }
+
+ return streams;
+ }
+ };
+
+ class PictureIdObserver : public test::RtpRtcpObserver {
+ public:
+ PictureIdObserver()
+ : test::RtpRtcpObserver(kDefaultTimeoutMs), num_ssrcs_to_observe_(1) {}
+
+ void ResetExpectations(size_t num_expected_ssrcs) {
+ rtc::CritScope lock(&crit_);
+ // Do not clear the timestamp and picture_id, to ensure that we check
+ // consistency between reinits and recreations.
+ num_packets_sent_.clear();
+ num_ssrcs_to_observe_ = num_expected_ssrcs;
+ ssrc_observed_.clear();
+ }
+
+ private:
+ Action OnSendRtp(const uint8_t* packet, size_t length) override {
+ rtc::CritScope lock(&crit_);
+
+ // RTP header.
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, length, &header));
+ const uint32_t timestamp = header.timestamp;
+ const uint32_t ssrc = header.ssrc;
+
+ const bool known_ssrc =
+ (ssrc == kVideoSendSsrcs[0] || ssrc == kVideoSendSsrcs[1] ||
+ ssrc == kVideoSendSsrcs[2]);
+ EXPECT_TRUE(known_ssrc) << "Unknown SSRC sent.";
+
+ const bool is_padding =
+ (length == header.headerLength + header.paddingLength);
+ if (is_padding) {
+ return SEND_PACKET;
+ }
+
+ // VP8 header.
+ std::unique_ptr<RtpDepacketizer> depacketizer(
+ RtpDepacketizer::Create(kRtpVideoVp8));
+ RtpDepacketizer::ParsedPayload parsed_payload;
+ EXPECT_TRUE(depacketizer->Parse(
+ &parsed_payload, &packet[header.headerLength],
+ length - header.headerLength - header.paddingLength));
+ const uint16_t picture_id =
+ parsed_payload.type.Video.codecHeader.VP8.pictureId;
+
+ // If this is the first packet, we have nothing to compare to.
+ if (last_observed_timestamp_.find(ssrc) ==
+ last_observed_timestamp_.end()) {
+ last_observed_timestamp_[ssrc] = timestamp;
+ last_observed_picture_id_[ssrc] = picture_id;
+ ++num_packets_sent_[ssrc];
+
+ return SEND_PACKET;
+ }
+
+ // Verify continuity and monotonicity of picture_id sequence.
+ if (last_observed_timestamp_[ssrc] == timestamp) {
+ // Packet belongs to same frame as before.
+ EXPECT_EQ(last_observed_picture_id_[ssrc], picture_id);
+ } else {
+ // Packet is a new frame.
+ EXPECT_EQ((last_observed_picture_id_[ssrc] + 1) % (1 << 15),
+ picture_id);
+ }
+ last_observed_timestamp_[ssrc] = timestamp;
+ last_observed_picture_id_[ssrc] = picture_id;
+
+ // Pass the test when enough media packets have been received
+ // on all streams.
+ if (++num_packets_sent_[ssrc] >= 10 && !ssrc_observed_[ssrc]) {
+ ssrc_observed_[ssrc] = true;
+ if (--num_ssrcs_to_observe_ == 0) {
+ observation_complete_.Set();
+ }
+ }
+
+ return SEND_PACKET;
+ }
+
+ rtc::CriticalSection crit_;
+ std::map<uint32_t, uint32_t> last_observed_timestamp_ GUARDED_BY(crit_);
+ std::map<uint32_t, uint16_t> last_observed_picture_id_ GUARDED_BY(crit_);
+ std::map<uint32_t, size_t> num_packets_sent_ GUARDED_BY(crit_);
+ size_t num_ssrcs_to_observe_ GUARDED_BY(crit_);
+ std::map<uint32_t, bool> ssrc_observed_ GUARDED_BY(crit_);
+ } observer;
+
+ Call::Config config(event_log_.get());
+ CreateCalls(config, config);
+
+ test::PacketTransport send_transport(
+ sender_call_.get(), &observer, test::PacketTransport::kSender,
+ payload_type_map_, FakeNetworkPipe::Config());
+ test::PacketTransport receive_transport(
+ nullptr, &observer, test::PacketTransport::kReceiver, payload_type_map_,
+ FakeNetworkPipe::Config());
+ send_transport.SetReceiver(receiver_call_->Receiver());
+ receive_transport.SetReceiver(sender_call_->Receiver());
+
+ CreateSendConfig(kNumSsrcs, 0, 0, &send_transport);
+ video_send_config_.encoder_settings.encoder = encoder;
+ video_send_config_.encoder_settings.payload_name = "VP8";
+ video_encoder_config_.video_stream_factory =
+ new rtc::RefCountedObject<VideoStreamFactory>();
+ video_encoder_config_.number_of_streams = 1;
+ CreateMatchingReceiveConfigs(&receive_transport);
+
+ CreateVideoStreams();
+ CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
+
+ auto reinit_encoder_and_test = [this, &observer](int num_expected_ssrcs) {
+ video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
+ observer.ResetExpectations(num_expected_ssrcs);
+ EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
+ };
+
+ // TODO(brandtr): Add tests where we recreate the whole VideoSendStream. This
+ // requires synchronizing the frame generator output with the packetization
+ // output, to not have any timing-dependent gaps in the picture_id sequence.
+
+ // Initial test with a single stream.
+ Start();
+ EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
+
+ // Reinit the encoder and make sure the picture_id sequence is continuous.
+ reinit_encoder_and_test(1);
+
+ // Go up to three streams.
+ video_encoder_config_.number_of_streams = 3;
+ reinit_encoder_and_test(3);
+ reinit_encoder_and_test(3);
+
+ // Go back to one stream.
+ video_encoder_config_.number_of_streams = 1;
+ reinit_encoder_and_test(1);
+ reinit_encoder_and_test(1);
+
+ send_transport.StopSending();
+ receive_transport.StopSending();
+
+ Stop();
+ DestroyStreams();
+}
+
+// These tests exposed a race in libvpx, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=7663. Disabling the tests
+// on tsan until the race has been fixed.
+#if defined(THREAD_SANITIZER)
+#define MAYBE_PictureIdStateRetainedAfterReinitingVp8 \
+ DISABLED_PictureIdStateRetainedAfterReinitingVp8
+#define MAYBE_PictureIdStateRetainedAfterReinitingSimulcastEncoderAdapter \
+ DISABLED_PictureIdStateRetainedAfterReinitingSimulcastEncoderAdapter
+#else
+#define MAYBE_PictureIdStateRetainedAfterReinitingVp8 \
+ PictureIdStateRetainedAfterReinitingVp8
+#define MAYBE_PictureIdStateRetainedAfterReinitingSimulcastEncoderAdapter \
+ PictureIdStateRetainedAfterReinitingSimulcastEncoderAdapter
+#endif
+TEST_F(EndToEndTest, MAYBE_PictureIdStateRetainedAfterReinitingVp8) {
+ std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
+ TestPictureIdStatePreservation(encoder.get());
+}
+
+TEST_F(EndToEndTest,
+ MAYBE_PictureIdStateRetainedAfterReinitingSimulcastEncoderAdapter) {
+ class VideoEncoderFactoryAdapter : public webrtc::VideoEncoderFactory {
+ public:
+ explicit VideoEncoderFactoryAdapter(
+ cricket::WebRtcVideoEncoderFactory* factory)
+ : factory_(factory) {}
+ virtual ~VideoEncoderFactoryAdapter() {}
+
+ // Implements webrtc::VideoEncoderFactory.
+ webrtc::VideoEncoder* Create() override {
+ return factory_->CreateVideoEncoder(
+ cricket::VideoCodec(cricket::kVp8CodecName));
+ }
+
+ void Destroy(webrtc::VideoEncoder* encoder) override {
+ return factory_->DestroyVideoEncoder(encoder);
+ }
+
+ private:
+ cricket::WebRtcVideoEncoderFactory* const factory_;
+ };
+
+ cricket::InternalEncoderFactory internal_encoder_factory;
+ SimulcastEncoderAdapter simulcast_encoder_adapter(
+ new VideoEncoderFactoryAdapter(&internal_encoder_factory));
+
+ TestPictureIdStatePreservation(&simulcast_encoder_adapter);
+}
+
TEST_F(EndToEndTest, RespectsNetworkState) {
// TODO(pbos): Remove accepted downtime packets etc. when signaling network
// down blocks until no more packets will be sent.
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