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Unified Diff: webrtc/video/rtp_stream_receiver_unittest.cc

Issue 2641463002: Unit test out of band H264 SPS,PPS within RtpStreamReceiver. (Closed)
Patch Set: Rebase. Created 3 years, 11 months ago
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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,
+ void(video_coding::FrameObject* frame));
+ MOCK_METHOD1(DoOnCompleteFrameFailLength,
+ 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) {
+ 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*.
+ 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")) {}
+
+ void SetUp() {
+ rtp_stream_receiver_.reset(new RtpStreamReceiver(
+ nullptr, nullptr, &mock_transport_, nullptr, &packet_router_,
+ nullptr, &config_, nullptr, process_thread_.get(),
+ &mock_nack_sender_, &mock_key_frame_request_sender_,
+ &mock_on_complete_frame_callback_, &timing_));
+ }
+
+ WebRtcRTPHeader GetDefaultPacket() {
+ WebRtcRTPHeader packet;
+ memset(&packet, 0, sizeof(packet));
+ packet.type.Video.codec = kRtpVideoH264;
+ return packet;
+ }
+
+ // TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate
+ // code.
+ void AddSps(WebRtcRTPHeader* packet, int sps_id, std::vector<uint8_t>* data) {
+ NaluInfo info;
+ info.type = H264::NaluType::kSps;
+ info.sps_id = sps_id;
+ info.pps_id = -1;
+ info.offset = data->size();
+ info.size = 2;
+ data->push_back(H264::NaluType::kSps);
+ data->push_back(sps_id);
+ packet->type.Video.codecHeader.H264
+ .nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
+ }
+
+ 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_;
+ MockOnCompleteFrameCallback mock_on_complete_frame_callback_;
+ PacketRouter packet_router_;
+ VCMTiming timing_;
+ std::unique_ptr<ProcessThread> process_thread_;
+ 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;
+ 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;
+ 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(),
+ &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
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