6c2c13af06
This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266. Reason for revert: Regression in ramp up perf tests. Original change's description: > Reland "Move rtp-specific config out of EncoderSettings." > > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c > > Original change's description: > > Move rtp-specific config out of EncoderSettings. > > > > In VideoSendStream::Config, move payload_name and payload_type from > > EncoderSettings to Rtp. > > > > EncoderSettings now contains configuration for VideoStreamEncoder only, > > and should perhaps be renamed in a follow up cl. It's no longer > > passed as an argument to VideoCodecInitializer::SetupCodec. > > > > The latter then needs a different way to know the codec type, > > which is provided by a new codec_type member in VideoEncoderConfig. > > > > Bug: webrtc:8830 > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6 > > Reviewed-on: https://webrtc-review.googlesource.com/62062 > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22532} > > Bug: webrtc:8830 > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019 > Reviewed-on: https://webrtc-review.googlesource.com/63721 > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22595} TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org Bug: webrtc:8830,chromium:827080 Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef Reviewed-on: https://webrtc-review.googlesource.com/65520 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22677}
81 lines
2.7 KiB
C++
81 lines
2.7 KiB
C++
/*
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* Copyright (c) 2012 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 "video/encoder_rtcp_feedback.h"
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#include <memory>
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include "video/send_statistics_proxy.h"
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#include "video/video_stream_encoder.h"
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using ::testing::NiceMock;
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namespace webrtc {
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class MockVideoStreamEncoder : public VideoStreamEncoder {
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public:
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explicit MockVideoStreamEncoder(SendStatisticsProxy* send_stats_proxy)
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: VideoStreamEncoder(1,
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send_stats_proxy,
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VideoSendStream::Config::EncoderSettings("fake", 0,
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nullptr),
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nullptr,
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rtc::MakeUnique<OveruseFrameDetector>(
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CpuOveruseOptions(), nullptr)) {}
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~MockVideoStreamEncoder() { Stop(); }
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MOCK_METHOD0(SendKeyFrame, void());
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};
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class VieKeyRequestTest : public ::testing::Test {
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public:
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VieKeyRequestTest()
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: simulated_clock_(123456789),
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send_stats_proxy_(&simulated_clock_,
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VideoSendStream::Config(nullptr),
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VideoEncoderConfig::ContentType::kRealtimeVideo),
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encoder_(&send_stats_proxy_),
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encoder_rtcp_feedback_(
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&simulated_clock_,
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std::vector<uint32_t>(1, VieKeyRequestTest::kSsrc),
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&encoder_) {}
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protected:
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const uint32_t kSsrc = 1234;
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SimulatedClock simulated_clock_;
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SendStatisticsProxy send_stats_proxy_;
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MockVideoStreamEncoder encoder_;
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EncoderRtcpFeedback encoder_rtcp_feedback_;
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};
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TEST_F(VieKeyRequestTest, CreateAndTriggerRequests) {
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EXPECT_CALL(encoder_, SendKeyFrame()).Times(1);
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encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
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}
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TEST_F(VieKeyRequestTest, TooManyOnReceivedIntraFrameRequest) {
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EXPECT_CALL(encoder_, SendKeyFrame()).Times(1);
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encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
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encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
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simulated_clock_.AdvanceTimeMilliseconds(10);
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encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
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EXPECT_CALL(encoder_, SendKeyFrame()).Times(1);
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simulated_clock_.AdvanceTimeMilliseconds(300);
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encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
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encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
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encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
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}
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} // namespace webrtc
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