Unit test out of band H264 SPS,PPS within RtpStreamReceiver.
This CL introduces a dedicated unit test for webrtc::RtpStreamReceiver. Focus of this CL is testing RtpStreamReciver::OnReceivedPayloadData(). Dependencies with virtual interfaces are (g)mocked, non-virtual dependencies are instantiated. This CL is chained to https://codereview.webrtc.org/2638933002/ . BUG=webrtc:5948 Review-Url: https://codereview.webrtc.org/2641463002 Cr-Commit-Position: refs/heads/master@{#16240}
This commit is contained in:
johan
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@ -29,6 +29,7 @@ namespace webrtc {
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bool H264SpropParameterSets::DecodeSprop(const std::string& sprop) {
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size_t separator_pos = sprop.find(',');
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LOG(LS_INFO) << "Parsing sprop \"" << sprop << "\"";
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if ((separator_pos <= 0) || (separator_pos >= sprop.length() - 1)) {
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LOG(LS_WARNING) << "Invalid seperator position " << separator_pos << " *"
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<< sprop << "*";
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@ -198,6 +198,7 @@ if (rtc_include_tests) {
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"quality_threshold_unittest.cc",
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"receive_statistics_proxy_unittest.cc",
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"report_block_stats_unittest.cc",
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"rtp_stream_receiver_unittest.cc",
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"send_delay_stats_unittest.cc",
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"send_statistics_proxy_unittest.cc",
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"stats_counter_unittest.cc",
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@ -213,7 +213,9 @@ RtpStreamReceiver::~RtpStreamReceiver() {
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packet_router_->RemoveRtpModule(rtp_rtcp_.get());
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rtp_rtcp_->SetREMBStatus(false);
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remb_->RemoveReceiveChannel(rtp_rtcp_.get());
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if (config_.rtp.remb) {
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remb_->RemoveReceiveChannel(rtp_rtcp_.get());
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}
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UpdateHistograms();
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}
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@ -249,7 +251,6 @@ int32_t RtpStreamReceiver::OnReceivedPayloadData(
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const uint8_t* payload_data,
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size_t payload_size,
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const WebRtcRTPHeader* rtp_header) {
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RTC_DCHECK(video_receiver_);
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WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
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rtp_header_with_ntp.ntp_time_ms =
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ntp_estimator_.Estimate(rtp_header->header.timestamp);
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@ -284,6 +285,7 @@ int32_t RtpStreamReceiver::OnReceivedPayloadData(
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packet_buffer_->InsertPacket(&packet);
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} else {
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RTC_DCHECK(video_receiver_);
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if (video_receiver_->IncomingPacket(payload_data, payload_size,
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rtp_header_with_ntp) != 0) {
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// Check this...
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@ -664,7 +666,7 @@ void RtpStreamReceiver::InsertSpsPpsIntoTracker(uint8_t payload_type) {
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return;
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LOG(LS_INFO) << "Found out of band supplied codec parameters for"
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<< " payload type: " << payload_type;
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<< " payload type: " << static_cast<int>(payload_type);
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H264SpropParameterSets sprop_decoder;
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auto sprop_base64_it =
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@ -673,7 +675,7 @@ void RtpStreamReceiver::InsertSpsPpsIntoTracker(uint8_t payload_type) {
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if (sprop_base64_it == codec_params_it->second.end())
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return;
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if (!sprop_decoder.DecodeSprop(sprop_base64_it->second))
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if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
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return;
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tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
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299
webrtc/video/rtp_stream_receiver_unittest.cc
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299
webrtc/video/rtp_stream_receiver_unittest.cc
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@ -0,0 +1,299 @@
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/*
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* Copyright 2017 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/test/gtest.h"
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#include "webrtc/test/gmock.h"
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#include "webrtc/base/bytebuffer.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/common_video/h264/h264_common.h"
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#include "webrtc/media/base/mediaconstants.h"
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#include "webrtc/modules/pacing/packet_router.h"
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#include "webrtc/modules/video_coding/include/video_coding_defines.h"
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#include "webrtc/modules/video_coding/frame_object.h"
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#include "webrtc/modules/video_coding/packet.h"
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#include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
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#include "webrtc/modules/video_coding/timing.h"
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#include "webrtc/modules/utility/include/process_thread.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/system_wrappers/include/field_trial_default.h"
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#include "webrtc/test/field_trial.h"
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#include "webrtc/video/rtp_stream_receiver.h"
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using testing::_;
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namespace webrtc {
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namespace {
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const char kNewJitterBufferFieldTrialEnabled[] =
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"WebRTC-NewVideoJitterBuffer/Enabled/";
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const uint8_t kH264StartCode[] = {0x00, 0x00, 0x00, 0x01};
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class MockTransport : public Transport {
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public:
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MOCK_METHOD3(SendRtp,
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bool(const uint8_t* packet,
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size_t length,
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const PacketOptions& options));
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MOCK_METHOD2(SendRtcp, bool(const uint8_t* packet, size_t length));
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};
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class MockNackSender : public NackSender {
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public:
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MOCK_METHOD1(SendNack, void(const std::vector<uint16_t>& sequence_numbers));
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};
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class MockKeyFrameRequestSender : public KeyFrameRequestSender {
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public:
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MOCK_METHOD0(RequestKeyFrame, void());
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};
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class MockOnCompleteFrameCallback
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: public video_coding::OnCompleteFrameCallback {
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public:
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MockOnCompleteFrameCallback() : buffer_(rtc::ByteBuffer::ORDER_NETWORK) {}
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MOCK_METHOD1(DoOnCompleteFrame, void(video_coding::FrameObject* frame));
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MOCK_METHOD1(DoOnCompleteFrameFailNullptr,
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void(video_coding::FrameObject* frame));
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MOCK_METHOD1(DoOnCompleteFrameFailLength,
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void(video_coding::FrameObject* frame));
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MOCK_METHOD1(DoOnCompleteFrameFailBitstream,
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void(video_coding::FrameObject* frame));
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void OnCompleteFrame(std::unique_ptr<video_coding::FrameObject> frame) {
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if (!frame) {
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DoOnCompleteFrameFailNullptr(nullptr);
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return;
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}
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EXPECT_EQ(buffer_.Length(), frame->size());
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if (buffer_.Length() != frame->size()) {
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DoOnCompleteFrameFailLength(frame.get());
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return;
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}
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std::vector<uint8_t> actual_data(frame->size());
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frame->GetBitstream(actual_data.data());
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if (memcmp(buffer_.Data(), actual_data.data(), buffer_.Length()) != 0) {
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DoOnCompleteFrameFailBitstream(frame.get());
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return;
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}
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DoOnCompleteFrame(frame.get());
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}
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void AppendExpectedBitstream(const uint8_t data[], size_t size_in_bytes) {
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// 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);
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}
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rtc::ByteBufferWriter buffer_;
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};
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} // namespace
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class RtpStreamReceiverTest : public testing::Test {
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public:
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RtpStreamReceiverTest()
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: config_(CreateConfig()),
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timing_(Clock::GetRealTimeClock()),
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process_thread_(ProcessThread::Create("TestThread")) {}
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void SetUp() {
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rtp_stream_receiver_.reset(new RtpStreamReceiver(
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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_));
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}
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WebRtcRTPHeader GetDefaultPacket() {
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WebRtcRTPHeader packet;
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memset(&packet, 0, sizeof(packet));
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packet.type.Video.codec = kRtpVideoH264;
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return packet;
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}
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// TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate
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// code.
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void AddSps(WebRtcRTPHeader* packet, int sps_id, std::vector<uint8_t>* data) {
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NaluInfo info;
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info.type = H264::NaluType::kSps;
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info.sps_id = sps_id;
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info.pps_id = -1;
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info.offset = data->size();
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info.size = 2;
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data->push_back(H264::NaluType::kSps);
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data->push_back(sps_id);
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packet->type.Video.codecHeader.H264
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.nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
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}
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void AddPps(WebRtcRTPHeader* packet,
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int sps_id,
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int pps_id,
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std::vector<uint8_t>* data) {
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NaluInfo info;
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info.type = H264::NaluType::kPps;
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info.sps_id = sps_id;
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info.pps_id = pps_id;
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info.offset = data->size();
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info.size = 2;
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data->push_back(H264::NaluType::kPps);
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data->push_back(pps_id);
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packet->type.Video.codecHeader.H264
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.nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
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}
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void AddIdr(WebRtcRTPHeader* packet, int pps_id) {
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NaluInfo info;
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info.type = H264::NaluType::kIdr;
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info.sps_id = -1;
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info.pps_id = pps_id;
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packet->type.Video.codecHeader.H264
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.nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
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}
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protected:
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static VideoReceiveStream::Config CreateConfig() {
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VideoReceiveStream::Config config(nullptr);
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config.rtp.remote_ssrc = 1111;
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config.rtp.local_ssrc = 2222;
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return config;
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}
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webrtc::test::ScopedFieldTrials override_field_trials_{
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kNewJitterBufferFieldTrialEnabled};
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VideoReceiveStream::Config config_;
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MockNackSender mock_nack_sender_;
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MockKeyFrameRequestSender mock_key_frame_request_sender_;
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MockTransport mock_transport_;
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MockOnCompleteFrameCallback mock_on_complete_frame_callback_;
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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_;
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};
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TEST_F(RtpStreamReceiverTest, GenericKeyFrame) {
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WebRtcRTPHeader rtp_header;
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const std::vector<uint8_t> data({1, 2, 3, 4});
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memset(&rtp_header, 0, sizeof(rtp_header));
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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;
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rtp_header.frameType = kVideoFrameKey;
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rtp_header.type.Video.codec = kRtpVideoGeneric;
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mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
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data.size());
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EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
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rtp_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
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&rtp_header);
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}
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TEST_F(RtpStreamReceiverTest, GenericKeyFrameBitstreamError) {
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WebRtcRTPHeader rtp_header;
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const std::vector<uint8_t> data({1, 2, 3, 4});
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memset(&rtp_header, 0, sizeof(rtp_header));
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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;
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rtp_header.frameType = kVideoFrameKey;
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rtp_header.type.Video.codec = kRtpVideoGeneric;
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constexpr uint8_t expected_bitsteam[] = {1, 2, 3, 0xff};
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mock_on_complete_frame_callback_.AppendExpectedBitstream(
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expected_bitsteam, sizeof(expected_bitsteam));
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EXPECT_CALL(mock_on_complete_frame_callback_,
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DoOnCompleteFrameFailBitstream(_));
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rtp_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
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&rtp_header);
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}
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TEST_F(RtpStreamReceiverTest, InBandSpsPps) {
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std::vector<uint8_t> sps_data;
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WebRtcRTPHeader sps_packet = GetDefaultPacket();
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AddSps(&sps_packet, 0, &sps_data);
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sps_packet.header.sequenceNumber = 0;
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mock_on_complete_frame_callback_.AppendExpectedBitstream(
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kH264StartCode, sizeof(kH264StartCode));
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mock_on_complete_frame_callback_.AppendExpectedBitstream(sps_data.data(),
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sps_data.size());
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rtp_stream_receiver_->OnReceivedPayloadData(sps_data.data(), sps_data.size(),
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&sps_packet);
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std::vector<uint8_t> pps_data;
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WebRtcRTPHeader pps_packet = GetDefaultPacket();
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AddPps(&pps_packet, 0, 1, &pps_data);
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pps_packet.header.sequenceNumber = 1;
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mock_on_complete_frame_callback_.AppendExpectedBitstream(
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kH264StartCode, sizeof(kH264StartCode));
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mock_on_complete_frame_callback_.AppendExpectedBitstream(pps_data.data(),
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pps_data.size());
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rtp_stream_receiver_->OnReceivedPayloadData(pps_data.data(), pps_data.size(),
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&pps_packet);
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std::vector<uint8_t> idr_data;
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WebRtcRTPHeader idr_packet = GetDefaultPacket();
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AddIdr(&idr_packet, 1);
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idr_packet.type.Video.is_first_packet_in_frame = true;
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idr_packet.header.sequenceNumber = 2;
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idr_packet.header.markerBit = 1;
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idr_packet.type.Video.is_first_packet_in_frame = true;
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idr_packet.frameType = kVideoFrameKey;
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idr_packet.type.Video.codec = kRtpVideoH264;
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idr_data.insert(idr_data.end(), {0x65, 1, 2, 3});
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mock_on_complete_frame_callback_.AppendExpectedBitstream(
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kH264StartCode, sizeof(kH264StartCode));
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mock_on_complete_frame_callback_.AppendExpectedBitstream(idr_data.data(),
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idr_data.size());
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EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
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rtp_stream_receiver_->OnReceivedPayloadData(idr_data.data(), idr_data.size(),
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&idr_packet);
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}
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TEST_F(RtpStreamReceiverTest, OutOfBandFmtpSpsPps) {
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constexpr int kPayloadType = 99;
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VideoCodec codec;
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codec.plType = kPayloadType;
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std::map<std::string, std::string> codec_params;
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// Example parameter sets from https://tools.ietf.org/html/rfc3984#section-8.2
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// .
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codec_params.insert(
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{cricket::kH264FmtpSpropParameterSets, "Z0IACpZTBYmI,aMljiA=="});
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rtp_stream_receiver_->AddReceiveCodec(codec, codec_params);
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const uint8_t binary_sps[] = {0x67, 0x42, 0x00, 0x0a, 0x96,
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0x53, 0x05, 0x89, 0x88};
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mock_on_complete_frame_callback_.AppendExpectedBitstream(
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kH264StartCode, sizeof(kH264StartCode));
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mock_on_complete_frame_callback_.AppendExpectedBitstream(binary_sps,
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sizeof(binary_sps));
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const uint8_t binary_pps[] = {0x68, 0xc9, 0x63, 0x88};
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mock_on_complete_frame_callback_.AppendExpectedBitstream(
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kH264StartCode, sizeof(kH264StartCode));
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mock_on_complete_frame_callback_.AppendExpectedBitstream(binary_pps,
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sizeof(binary_pps));
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std::vector<uint8_t> data;
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WebRtcRTPHeader idr_packet = GetDefaultPacket();
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AddIdr(&idr_packet, 0);
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idr_packet.header.payloadType = kPayloadType;
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idr_packet.type.Video.is_first_packet_in_frame = true;
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idr_packet.header.sequenceNumber = 2;
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idr_packet.header.markerBit = 1;
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idr_packet.type.Video.is_first_packet_in_frame = true;
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idr_packet.frameType = kVideoFrameKey;
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idr_packet.type.Video.codec = kRtpVideoH264;
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data.insert(data.end(), {1, 2, 3});
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mock_on_complete_frame_callback_.AppendExpectedBitstream(
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kH264StartCode, sizeof(kH264StartCode));
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mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
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data.size());
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EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
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rtp_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
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&idr_packet);
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}
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} // namespace webrtc
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