| Index: webrtc/video/rtp_stream_receiver_unittest.cc
|
| diff --git a/webrtc/video/rtp_stream_receiver_unittest.cc b/webrtc/video/rtp_stream_receiver_unittest.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..6573132466ad0bac4793c6b8c0436dcd210ba572
|
| --- /dev/null
|
| +++ b/webrtc/video/rtp_stream_receiver_unittest.cc
|
| @@ -0,0 +1,299 @@
|
| +/*
|
| + * 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 "webrtc/test/gtest.h"
|
| +#include "webrtc/test/gmock.h"
|
| +
|
| +#include "webrtc/base/bytebuffer.h"
|
| +#include "webrtc/base/logging.h"
|
| +#include "webrtc/common_video/h264/h264_common.h"
|
| +#include "webrtc/media/base/mediaconstants.h"
|
| +#include "webrtc/modules/pacing/packet_router.h"
|
| +#include "webrtc/modules/video_coding/include/video_coding_defines.h"
|
| +#include "webrtc/modules/video_coding/frame_object.h"
|
| +#include "webrtc/modules/video_coding/packet.h"
|
| +#include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
|
| +#include "webrtc/modules/video_coding/timing.h"
|
| +#include "webrtc/modules/utility/include/process_thread.h"
|
| +#include "webrtc/system_wrappers/include/clock.h"
|
| +#include "webrtc/system_wrappers/include/field_trial_default.h"
|
| +#include "webrtc/test/field_trial.h"
|
| +#include "webrtc/video/rtp_stream_receiver.h"
|
| +
|
| +using testing::_;
|
| +
|
| +namespace webrtc {
|
| +
|
| +namespace {
|
| +
|
| +const char kNewJitterBufferFieldTrialEnabled[] =
|
| + "WebRTC-NewVideoJitterBuffer/Enabled/";
|
| +const uint8_t kH264StartCode[] = {0x00, 0x00, 0x00, 0x01};
|
| +
|
| +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 MockNackSender : public NackSender {
|
| + public:
|
| + MOCK_METHOD1(SendNack, void(const std::vector<uint16_t>& sequence_numbers));
|
| +};
|
| +
|
| +class MockKeyFrameRequestSender : public KeyFrameRequestSender {
|
| + public:
|
| + MOCK_METHOD0(RequestKeyFrame, void());
|
| +};
|
| +
|
| +class MockOnCompleteFrameCallback
|
| + : public video_coding::OnCompleteFrameCallback {
|
| + public:
|
| + MockOnCompleteFrameCallback() : buffer_(rtc::ByteBuffer::ORDER_NETWORK) {}
|
| +
|
| + MOCK_METHOD1(DoOnCompleteFrame, void(video_coding::FrameObject* frame));
|
| + MOCK_METHOD1(DoOnCompleteFrameFailNullptr,
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| + void(video_coding::FrameObject* frame));
|
| + MOCK_METHOD1(DoOnCompleteFrameFailLength,
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| + void(video_coding::FrameObject* frame));
|
| + MOCK_METHOD1(DoOnCompleteFrameFailBitstream,
|
| + void(video_coding::FrameObject* frame));
|
| + void OnCompleteFrame(std::unique_ptr<video_coding::FrameObject> frame) {
|
| + if (!frame) {
|
| + DoOnCompleteFrameFailNullptr(nullptr);
|
| + return;
|
| + }
|
| + EXPECT_EQ(buffer_.Length(), frame->size());
|
| + if (buffer_.Length() != frame->size()) {
|
| + DoOnCompleteFrameFailLength(frame.get());
|
| + return;
|
| + }
|
| + std::vector<uint8_t> actual_data(frame->size());
|
| + frame->GetBitstream(actual_data.data());
|
| + if (memcmp(buffer_.Data(), actual_data.data(), buffer_.Length()) != 0) {
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| + DoOnCompleteFrameFailBitstream(frame.get());
|
| + return;
|
| + }
|
| + DoOnCompleteFrame(frame.get());
|
| + }
|
| + void AppendExpectedBitstream(const uint8_t data[], size_t size_in_bytes) {
|
| + // TODO(Johan): Let rtc::ByteBuffer handle uint8_t* instead of char*.
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| + buffer_.WriteBytes(reinterpret_cast<const char*>(data), size_in_bytes);
|
| + }
|
| + rtc::ByteBufferWriter buffer_;
|
| +};
|
| +
|
| +} // namespace
|
| +
|
| +class RtpStreamReceiverTest : public testing::Test {
|
| + public:
|
| + RtpStreamReceiverTest()
|
| + : config_(CreateConfig()),
|
| + timing_(Clock::GetRealTimeClock()),
|
| + process_thread_(ProcessThread::Create("TestThread")) {}
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| +
|
| + void SetUp() {
|
| + rtp_stream_receiver_.reset(new RtpStreamReceiver(
|
| + nullptr, nullptr, &mock_transport_, nullptr, &packet_router_,
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| + nullptr, &config_, nullptr, process_thread_.get(),
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| + &mock_nack_sender_, &mock_key_frame_request_sender_,
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| + &mock_on_complete_frame_callback_, &timing_));
|
| + }
|
| +
|
| + WebRtcRTPHeader GetDefaultPacket() {
|
| + WebRtcRTPHeader packet;
|
| + memset(&packet, 0, sizeof(packet));
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| + packet.type.Video.codec = kRtpVideoH264;
|
| + return packet;
|
| + }
|
| +
|
| + // TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate
|
| + // code.
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| + void AddSps(WebRtcRTPHeader* packet, int sps_id, std::vector<uint8_t>* data) {
|
| + NaluInfo info;
|
| + info.type = H264::NaluType::kSps;
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| + info.sps_id = sps_id;
|
| + info.pps_id = -1;
|
| + info.offset = data->size();
|
| + info.size = 2;
|
| + data->push_back(H264::NaluType::kSps);
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| + data->push_back(sps_id);
|
| + packet->type.Video.codecHeader.H264
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| + .nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
|
| + }
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| +
|
| + void AddPps(WebRtcRTPHeader* packet,
|
| + int sps_id,
|
| + int pps_id,
|
| + std::vector<uint8_t>* data) {
|
| + NaluInfo info;
|
| + info.type = H264::NaluType::kPps;
|
| + info.sps_id = sps_id;
|
| + info.pps_id = pps_id;
|
| + info.offset = data->size();
|
| + info.size = 2;
|
| + data->push_back(H264::NaluType::kPps);
|
| + data->push_back(pps_id);
|
| + packet->type.Video.codecHeader.H264
|
| + .nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
|
| + }
|
| +
|
| + void AddIdr(WebRtcRTPHeader* packet, int pps_id) {
|
| + NaluInfo info;
|
| + info.type = H264::NaluType::kIdr;
|
| + info.sps_id = -1;
|
| + info.pps_id = pps_id;
|
| + packet->type.Video.codecHeader.H264
|
| + .nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
|
| + }
|
| +
|
| + protected:
|
| + static VideoReceiveStream::Config CreateConfig() {
|
| + VideoReceiveStream::Config config(nullptr);
|
| + config.rtp.remote_ssrc = 1111;
|
| + config.rtp.local_ssrc = 2222;
|
| + return config;
|
| + }
|
| +
|
| + webrtc::test::ScopedFieldTrials override_field_trials_{
|
| + kNewJitterBufferFieldTrialEnabled};
|
| + VideoReceiveStream::Config config_;
|
| + MockNackSender mock_nack_sender_;
|
| + MockKeyFrameRequestSender mock_key_frame_request_sender_;
|
| + MockTransport mock_transport_;
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| + MockOnCompleteFrameCallback mock_on_complete_frame_callback_;
|
| + PacketRouter packet_router_;
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| + VCMTiming timing_;
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| + std::unique_ptr<ProcessThread> process_thread_;
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| + std::unique_ptr<RtpStreamReceiver> rtp_stream_receiver_;
|
| +};
|
| +
|
| +TEST_F(RtpStreamReceiverTest, GenericKeyFrame) {
|
| + WebRtcRTPHeader rtp_header;
|
| + const std::vector<uint8_t> data({1, 2, 3, 4});
|
| + memset(&rtp_header, 0, sizeof(rtp_header));
|
| + rtp_header.header.sequenceNumber = 1;
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| + rtp_header.header.markerBit = 1;
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| + rtp_header.type.Video.is_first_packet_in_frame = true;
|
| + rtp_header.frameType = kVideoFrameKey;
|
| + rtp_header.type.Video.codec = kRtpVideoGeneric;
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
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| + data.size());
|
| + EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
|
| + rtp_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
| + &rtp_header);
|
| +}
|
| +
|
| +TEST_F(RtpStreamReceiverTest, GenericKeyFrameBitstreamError) {
|
| + WebRtcRTPHeader rtp_header;
|
| + const std::vector<uint8_t> data({1, 2, 3, 4});
|
| + memset(&rtp_header, 0, sizeof(rtp_header));
|
| + rtp_header.header.sequenceNumber = 1;
|
| + rtp_header.header.markerBit = 1;
|
| + rtp_header.type.Video.is_first_packet_in_frame = true;
|
| + rtp_header.frameType = kVideoFrameKey;
|
| + rtp_header.type.Video.codec = kRtpVideoGeneric;
|
| + constexpr uint8_t expected_bitsteam[] = {1, 2, 3, 0xff};
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
| + expected_bitsteam, sizeof(expected_bitsteam));
|
| + EXPECT_CALL(mock_on_complete_frame_callback_,
|
| + DoOnCompleteFrameFailBitstream(_));
|
| + rtp_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
| + &rtp_header);
|
| +}
|
| +
|
| +TEST_F(RtpStreamReceiverTest, InBandSpsPps) {
|
| + std::vector<uint8_t> sps_data;
|
| + WebRtcRTPHeader sps_packet = GetDefaultPacket();
|
| + AddSps(&sps_packet, 0, &sps_data);
|
| + sps_packet.header.sequenceNumber = 0;
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
| + kH264StartCode, sizeof(kH264StartCode));
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(sps_data.data(),
|
| + sps_data.size());
|
| + rtp_stream_receiver_->OnReceivedPayloadData(sps_data.data(), sps_data.size(),
|
| + &sps_packet);
|
| +
|
| + std::vector<uint8_t> pps_data;
|
| + WebRtcRTPHeader pps_packet = GetDefaultPacket();
|
| + AddPps(&pps_packet, 0, 1, &pps_data);
|
| + pps_packet.header.sequenceNumber = 1;
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
| + kH264StartCode, sizeof(kH264StartCode));
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(pps_data.data(),
|
| + pps_data.size());
|
| + rtp_stream_receiver_->OnReceivedPayloadData(pps_data.data(), pps_data.size(),
|
| + &pps_packet);
|
| +
|
| + std::vector<uint8_t> idr_data;
|
| + WebRtcRTPHeader idr_packet = GetDefaultPacket();
|
| + AddIdr(&idr_packet, 1);
|
| + idr_packet.type.Video.is_first_packet_in_frame = true;
|
| + idr_packet.header.sequenceNumber = 2;
|
| + idr_packet.header.markerBit = 1;
|
| + idr_packet.type.Video.is_first_packet_in_frame = true;
|
| + idr_packet.frameType = kVideoFrameKey;
|
| + idr_packet.type.Video.codec = kRtpVideoH264;
|
| + idr_data.insert(idr_data.end(), {0x65, 1, 2, 3});
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
| + kH264StartCode, sizeof(kH264StartCode));
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(idr_data.data(),
|
| + idr_data.size());
|
| + EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
|
| + rtp_stream_receiver_->OnReceivedPayloadData(idr_data.data(), idr_data.size(),
|
| + &idr_packet);
|
| +}
|
| +
|
| +TEST_F(RtpStreamReceiverTest, OutOfBandFmtpSpsPps) {
|
| + constexpr int kPayloadType = 99;
|
| + VideoCodec codec;
|
| + codec.plType = kPayloadType;
|
| + std::map<std::string, std::string> codec_params;
|
| + // Example parameter sets from https://tools.ietf.org/html/rfc3984#section-8.2
|
| + // .
|
| + codec_params.insert(
|
| + {cricket::kH264FmtpSpropParameterSets, "Z0IACpZTBYmI,aMljiA=="});
|
| + rtp_stream_receiver_->AddReceiveCodec(codec, codec_params);
|
| + const uint8_t binary_sps[] = {0x67, 0x42, 0x00, 0x0a, 0x96,
|
| + 0x53, 0x05, 0x89, 0x88};
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
| + kH264StartCode, sizeof(kH264StartCode));
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(binary_sps,
|
| + sizeof(binary_sps));
|
| + const uint8_t binary_pps[] = {0x68, 0xc9, 0x63, 0x88};
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
| + kH264StartCode, sizeof(kH264StartCode));
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(binary_pps,
|
| + sizeof(binary_pps));
|
| +
|
| + std::vector<uint8_t> data;
|
| + WebRtcRTPHeader idr_packet = GetDefaultPacket();
|
| + AddIdr(&idr_packet, 0);
|
| + idr_packet.header.payloadType = kPayloadType;
|
| + idr_packet.type.Video.is_first_packet_in_frame = true;
|
| + idr_packet.header.sequenceNumber = 2;
|
| + idr_packet.header.markerBit = 1;
|
| + idr_packet.type.Video.is_first_packet_in_frame = true;
|
| + idr_packet.frameType = kVideoFrameKey;
|
| + idr_packet.type.Video.codec = kRtpVideoH264;
|
| + data.insert(data.end(), {1, 2, 3});
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(
|
| + kH264StartCode, sizeof(kH264StartCode));
|
| + mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
|
| + data.size());
|
| + EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
|
| + rtp_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
|
| + &idr_packet);
|
| +}
|
| +
|
| +} // namespace webrtc
|
|
|