| Index: webrtc/video/video_receive_stream_unittest.cc
|
| diff --git a/webrtc/video/video_receive_stream_unittest.cc b/webrtc/video/video_receive_stream_unittest.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..6160e284f4c2a0ac9ddf401c0fa05a62310857d3
|
| --- /dev/null
|
| +++ b/webrtc/video/video_receive_stream_unittest.cc
|
| @@ -0,0 +1,141 @@
|
| +/*
|
| + * Copyright 2017 The WebRTC Project Authors. All rights reserved.
|
| + *
|
| + * Use of this source code is governed by a BSD-style license
|
| + * that can be found in the LICENSE file in the root of the source
|
| + * tree. An additional intellectual property rights grant can be found
|
| + * in the file PATENTS. All contributing project authors may
|
| + * be found in the AUTHORS file in the root of the source tree.
|
| + */
|
| +
|
| +#include <vector>
|
| +
|
| +#include "webrtc/test/gtest.h"
|
| +#include "webrtc/test/gmock.h"
|
| +
|
| +#include "webrtc/base/criticalsection.h"
|
| +#include "webrtc/base/event.h"
|
| +#include "webrtc/media/base/fakevideorenderer.h"
|
| +#include "webrtc/modules/pacing/packet_router.h"
|
| +#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
|
| +#include "webrtc/modules/utility/include/process_thread.h"
|
| +#include "webrtc/video/call_stats.h"
|
| +#include "webrtc/video/video_receive_stream.h"
|
| +#include "webrtc/system_wrappers/include/clock.h"
|
| +#include "webrtc/system_wrappers/include/sleep.h"
|
| +#include "webrtc/test/field_trial.h"
|
| +#include "webrtc/video_decoder.h"
|
| +
|
| +using testing::_;
|
| +using testing::Invoke;
|
| +
|
| +constexpr int kDefaultTimeOutMs = 50;
|
| +
|
| +namespace webrtc {
|
| +
|
| +namespace {
|
| +
|
| +const char kNewJitterBufferFieldTrialEnabled[] =
|
| + "WebRTC-NewVideoJitterBuffer/Enabled/";
|
| +
|
| +class MockTransport : public Transport {
|
| + public:
|
| + MOCK_METHOD3(SendRtp,
|
| + bool(const uint8_t* packet,
|
| + size_t length,
|
| + const PacketOptions& options));
|
| + MOCK_METHOD2(SendRtcp, bool(const uint8_t* packet, size_t length));
|
| +};
|
| +
|
| +class MockVideoDecoder : public VideoDecoder {
|
| + public:
|
| + MOCK_METHOD2(InitDecode,
|
| + int32_t(const VideoCodec* config, int32_t number_of_cores));
|
| + MOCK_METHOD5(Decode,
|
| + int32_t(const EncodedImage& input,
|
| + bool missing_frames,
|
| + const RTPFragmentationHeader* fragmentation,
|
| + const CodecSpecificInfo* codec_specific_info,
|
| + int64_t render_time_ms));
|
| + MOCK_METHOD1(RegisterDecodeCompleteCallback,
|
| + int32_t(DecodedImageCallback* callback));
|
| + MOCK_METHOD0(Release, int32_t(void));
|
| + const char* ImplementationName() const { return "MockVideoDecoder"; }
|
| +};
|
| +
|
| +} // namespace
|
| +
|
| +class VideoReceiveStreamTest : public testing::Test {
|
| + public:
|
| + VideoReceiveStreamTest()
|
| + : override_field_trials_(kNewJitterBufferFieldTrialEnabled),
|
| + config_(&mock_transport_),
|
| + call_stats_(Clock::GetRealTimeClock()),
|
| + process_thread_(ProcessThread::Create("TestThread")) {}
|
| +
|
| + void SetUp() {
|
| + constexpr int kDefaultNumCpuCores = 2;
|
| + config_.rtp.remote_ssrc = 1111;
|
| + config_.rtp.local_ssrc = 2222;
|
| + config_.renderer = &fake_renderer_;
|
| + VideoReceiveStream::Decoder h264_decoder;
|
| + h264_decoder.payload_type = 99;
|
| + h264_decoder.payload_name = "H264";
|
| + h264_decoder.codec_params.insert(
|
| + {"sprop-parameter-sets", "Z0IACpZTBYmI,aMljiA=="});
|
| + h264_decoder.decoder = &mock_h264_video_decoder_;
|
| + config_.decoders.push_back(h264_decoder);
|
| + VideoReceiveStream::Decoder null_decoder;
|
| + null_decoder.payload_type = 98;
|
| + null_decoder.payload_name = "null";
|
| + null_decoder.decoder = &mock_null_video_decoder_;
|
| + config_.decoders.push_back(null_decoder);
|
| +
|
| + video_receive_stream_.reset(new webrtc::internal::VideoReceiveStream(
|
| + kDefaultNumCpuCores,
|
| + false, // flex_fec
|
| + &packet_router_, config_.Copy(), process_thread_.get(), &call_stats_,
|
| + nullptr)); // remb
|
| + }
|
| +
|
| + protected:
|
| + webrtc::test::ScopedFieldTrials override_field_trials_;
|
| + VideoReceiveStream::Config config_;
|
| + CallStats call_stats_;
|
| + MockVideoDecoder mock_h264_video_decoder_;
|
| + MockVideoDecoder mock_null_video_decoder_;
|
| + cricket::FakeVideoRenderer fake_renderer_;
|
| + MockTransport mock_transport_;
|
| + PacketRouter packet_router_;
|
| + std::unique_ptr<ProcessThread> process_thread_;
|
| + std::unique_ptr<webrtc::internal::VideoReceiveStream> video_receive_stream_;
|
| +};
|
| +
|
| +TEST_F(VideoReceiveStreamTest, CreateFrameFromH264FmtpSpropAndIdr) {
|
| + constexpr uint8_t idr_nalu[] = {0x05, 0xFF, 0xFF, 0xFF};
|
| + RtpPacketToSend rtppacket(nullptr);
|
| + uint8_t* payload = rtppacket.AllocatePayload(sizeof(idr_nalu));
|
| + memcpy(payload, idr_nalu, sizeof(idr_nalu));
|
| + rtppacket.SetMarker(true);
|
| + rtppacket.SetSsrc(1111);
|
| + rtppacket.SetPayloadType(99);
|
| + rtppacket.SetSequenceNumber(1);
|
| + rtppacket.SetTimestamp(0);
|
| + rtc::Event init_decode_event_(false, false);
|
| + EXPECT_CALL(mock_h264_video_decoder_, InitDecode(_, _))
|
| + .WillOnce(Invoke([&init_decode_event_](const VideoCodec* config,
|
| + int32_t number_of_cores) {
|
| + init_decode_event_.Set();
|
| + return 0;
|
| + }));
|
| + EXPECT_CALL(mock_h264_video_decoder_, RegisterDecodeCompleteCallback(_));
|
| + video_receive_stream_->Start();
|
| + EXPECT_CALL(mock_h264_video_decoder_, Decode(_, false, _, _, _));
|
| + EXPECT_EQ(true,
|
| + video_receive_stream_->OnRecoveredPacket(rtppacket.data(),
|
| + rtppacket.size()));
|
| + EXPECT_CALL(mock_h264_video_decoder_, Release());
|
| + // Make sure the decoder thread had a chance to run.
|
| + init_decode_event_.Wait(kDefaultTimeOutMs);
|
| +}
|
| +} // namespace webrtc
|
|
|