blob: e76c3a9b401e1daaaad7898a8c1cc7a73e908190 [file] [log] [blame]
/*
* 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 "test/gtest.h"
#include "test/gmock.h"
#include "api/video_codecs/video_decoder.h"
#include "call/rtp_stream_receiver_controller.h"
#include "media/base/fakevideorenderer.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/event.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "video/call_stats.h"
#include "video/video_receive_stream.h"
namespace webrtc {
namespace {
using testing::_;
using testing::Invoke;
constexpr int kDefaultTimeOutMs = 50;
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()
: process_thread_(ProcessThread::Create("TestThread")),
override_field_trials_(kNewJitterBufferFieldTrialEnabled),
config_(&mock_transport_),
call_stats_(Clock::GetRealTimeClock(), process_thread_.get()) {}
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(
&rtp_stream_receiver_controller_, kDefaultNumCpuCores,
&packet_router_, config_.Copy(), process_thread_.get(), &call_stats_));
}
protected:
std::unique_ptr<ProcessThread> process_thread_;
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_;
RtpStreamReceiverController rtp_stream_receiver_controller_;
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, _, _, _));
RtpPacketReceived parsed_packet;
ASSERT_TRUE(parsed_packet.Parse(rtppacket.data(), rtppacket.size()));
rtp_stream_receiver_controller_.OnRtpPacket(parsed_packet);
EXPECT_CALL(mock_h264_video_decoder_, Release());
// Make sure the decoder thread had a chance to run.
init_decode_event_.Wait(kDefaultTimeOutMs);
}
} // namespace webrtc