
This reverts commit 00d0a0a1a9520fb4323d7e3a1c02133b7b942978. Reason for revert: Breaks downstream tests Original change's description: > Copy video frames metadata between encoded and plain frames in one place > > Currently some video frames metadata like rotation or ntp timestamps are > copied in every encoder and decoder separately. This CL makes copying to > happen at a single place for send or receive side. This will make it > easier to add new metadata in the future. > > Also, added some missing tests. > > Bug: webrtc:10460 > Change-Id: Ia49072c3041e75433f125a61050d2982b2bec1da > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346 > Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Johannes Kron <kron@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27719} TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org Change-Id: I8960a6cc15e552925129ba0037f197ff3fd93c25 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10460 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134100 Reviewed-by: Artem Titarenko <artit@webrtc.org> Commit-Queue: Artem Titarenko <artit@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27737}
216 lines
8.1 KiB
C++
216 lines
8.1 KiB
C++
/*
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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 <utility>
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#include <vector>
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "api/video_codecs/video_decoder.h"
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#include "call/rtp_stream_receiver_controller.h"
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#include "media/base/fake_video_renderer.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/event.h"
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#include "system_wrappers/include/clock.h"
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#include "test/field_trial.h"
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#include "test/video_decoder_proxy_factory.h"
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#include "video/call_stats.h"
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#include "video/video_receive_stream.h"
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namespace webrtc {
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namespace {
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using ::testing::_;
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using ::testing::Invoke;
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constexpr int kDefaultTimeOutMs = 50;
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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 MockVideoDecoder : public VideoDecoder {
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public:
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MOCK_METHOD2(InitDecode,
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int32_t(const VideoCodec* config, int32_t number_of_cores));
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MOCK_METHOD3(Decode,
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int32_t(const EncodedImage& input,
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bool missing_frames,
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int64_t render_time_ms));
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MOCK_METHOD1(RegisterDecodeCompleteCallback,
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int32_t(DecodedImageCallback* callback));
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MOCK_METHOD0(Release, int32_t(void));
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const char* ImplementationName() const { return "MockVideoDecoder"; }
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};
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class FrameObjectFake : public video_coding::EncodedFrame {
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public:
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int64_t ReceivedTime() const override { return 0; }
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int64_t RenderTime() const override { return _renderTimeMs; }
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};
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} // namespace
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class VideoReceiveStreamTest : public ::testing::Test {
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public:
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VideoReceiveStreamTest()
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: process_thread_(ProcessThread::Create("TestThread")),
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task_queue_factory_(CreateDefaultTaskQueueFactory()),
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config_(&mock_transport_),
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call_stats_(Clock::GetRealTimeClock(), process_thread_.get()),
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h264_decoder_factory_(&mock_h264_video_decoder_),
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null_decoder_factory_(&mock_null_video_decoder_) {}
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void SetUp() {
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constexpr int kDefaultNumCpuCores = 2;
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config_.rtp.remote_ssrc = 1111;
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config_.rtp.local_ssrc = 2222;
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config_.renderer = &fake_renderer_;
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VideoReceiveStream::Decoder h264_decoder;
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h264_decoder.payload_type = 99;
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h264_decoder.video_format = SdpVideoFormat("H264");
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h264_decoder.video_format.parameters.insert(
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{"sprop-parameter-sets", "Z0IACpZTBYmI,aMljiA=="});
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h264_decoder.decoder_factory = &h264_decoder_factory_;
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config_.decoders.push_back(h264_decoder);
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VideoReceiveStream::Decoder null_decoder;
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null_decoder.payload_type = 98;
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null_decoder.video_format = SdpVideoFormat("null");
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null_decoder.decoder_factory = &null_decoder_factory_;
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config_.decoders.push_back(null_decoder);
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Clock* clock = Clock::GetRealTimeClock();
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timing_ = new VCMTiming(clock);
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video_receive_stream_.reset(new webrtc::internal::VideoReceiveStream(
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task_queue_factory_.get(), &rtp_stream_receiver_controller_,
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kDefaultNumCpuCores, &packet_router_, config_.Copy(),
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process_thread_.get(), &call_stats_, clock, timing_));
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}
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protected:
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std::unique_ptr<ProcessThread> process_thread_;
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const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
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VideoReceiveStream::Config config_;
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CallStats call_stats_;
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MockVideoDecoder mock_h264_video_decoder_;
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MockVideoDecoder mock_null_video_decoder_;
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test::VideoDecoderProxyFactory h264_decoder_factory_;
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test::VideoDecoderProxyFactory null_decoder_factory_;
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cricket::FakeVideoRenderer fake_renderer_;
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MockTransport mock_transport_;
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PacketRouter packet_router_;
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RtpStreamReceiverController rtp_stream_receiver_controller_;
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std::unique_ptr<webrtc::internal::VideoReceiveStream> video_receive_stream_;
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VCMTiming* timing_;
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};
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TEST_F(VideoReceiveStreamTest, CreateFrameFromH264FmtpSpropAndIdr) {
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constexpr uint8_t idr_nalu[] = {0x05, 0xFF, 0xFF, 0xFF};
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RtpPacketToSend rtppacket(nullptr);
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uint8_t* payload = rtppacket.AllocatePayload(sizeof(idr_nalu));
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memcpy(payload, idr_nalu, sizeof(idr_nalu));
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rtppacket.SetMarker(true);
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rtppacket.SetSsrc(1111);
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rtppacket.SetPayloadType(99);
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rtppacket.SetSequenceNumber(1);
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rtppacket.SetTimestamp(0);
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rtc::Event init_decode_event_;
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EXPECT_CALL(mock_h264_video_decoder_, InitDecode(_, _))
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.WillOnce(Invoke([&init_decode_event_](const VideoCodec* config,
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int32_t number_of_cores) {
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init_decode_event_.Set();
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return 0;
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}));
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EXPECT_CALL(mock_h264_video_decoder_, RegisterDecodeCompleteCallback(_));
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video_receive_stream_->Start();
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EXPECT_CALL(mock_h264_video_decoder_, Decode(_, false, _));
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RtpPacketReceived parsed_packet;
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ASSERT_TRUE(parsed_packet.Parse(rtppacket.data(), rtppacket.size()));
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rtp_stream_receiver_controller_.OnRtpPacket(parsed_packet);
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EXPECT_CALL(mock_h264_video_decoder_, Release());
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// Make sure the decoder thread had a chance to run.
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init_decode_event_.Wait(kDefaultTimeOutMs);
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}
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TEST_F(VideoReceiveStreamTest, PlayoutDelay) {
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const PlayoutDelay kPlayoutDelayMs = {123, 321};
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std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
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test_frame->id.picture_id = 0;
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test_frame->SetPlayoutDelay(kPlayoutDelayMs);
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video_receive_stream_->OnCompleteFrame(std::move(test_frame));
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EXPECT_EQ(kPlayoutDelayMs.min_ms, timing_->min_playout_delay());
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EXPECT_EQ(kPlayoutDelayMs.max_ms, timing_->max_playout_delay());
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// Check that the biggest minimum delay is chosen.
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video_receive_stream_->SetMinimumPlayoutDelay(400);
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EXPECT_EQ(400, timing_->min_playout_delay());
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// Check base minimum delay validation.
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EXPECT_FALSE(video_receive_stream_->SetBaseMinimumPlayoutDelayMs(12345));
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EXPECT_FALSE(video_receive_stream_->SetBaseMinimumPlayoutDelayMs(-1));
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EXPECT_TRUE(video_receive_stream_->SetBaseMinimumPlayoutDelayMs(500));
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EXPECT_EQ(500, timing_->min_playout_delay());
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// Check that intermidiate values are remembered and the biggest remembered
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// is chosen.
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video_receive_stream_->SetBaseMinimumPlayoutDelayMs(0);
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EXPECT_EQ(400, timing_->min_playout_delay());
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video_receive_stream_->SetMinimumPlayoutDelay(0);
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EXPECT_EQ(123, timing_->min_playout_delay());
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}
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TEST_F(VideoReceiveStreamTest, PlayoutDelayPreservesDefaultMaxValue) {
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const int default_max_playout_latency = timing_->max_playout_delay();
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const PlayoutDelay kPlayoutDelayMs = {123, -1};
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std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
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test_frame->id.picture_id = 0;
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test_frame->SetPlayoutDelay(kPlayoutDelayMs);
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video_receive_stream_->OnCompleteFrame(std::move(test_frame));
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// Ensure that -1 preserves default maximum value from |timing_|.
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EXPECT_EQ(kPlayoutDelayMs.min_ms, timing_->min_playout_delay());
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EXPECT_NE(kPlayoutDelayMs.max_ms, timing_->max_playout_delay());
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EXPECT_EQ(default_max_playout_latency, timing_->max_playout_delay());
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}
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TEST_F(VideoReceiveStreamTest, PlayoutDelayPreservesDefaultMinValue) {
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const int default_min_playout_latency = timing_->min_playout_delay();
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const PlayoutDelay kPlayoutDelayMs = {-1, 321};
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std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
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test_frame->id.picture_id = 0;
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test_frame->SetPlayoutDelay(kPlayoutDelayMs);
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video_receive_stream_->OnCompleteFrame(std::move(test_frame));
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// Ensure that -1 preserves default minimum value from |timing_|.
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EXPECT_NE(kPlayoutDelayMs.min_ms, timing_->min_playout_delay());
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EXPECT_EQ(kPlayoutDelayMs.max_ms, timing_->max_playout_delay());
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EXPECT_EQ(default_min_playout_latency, timing_->min_playout_delay());
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}
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} // namespace webrtc
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