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This reverts commit 2666cf7eba
.
Reason for revert: On Lollipop Nexus 4, the 240p tests fail too.
Original change's description:
> Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld).
>
> * Add audio_ prefix to CallTest::{en,de}coder_factory_.
> * Let VideoQualityTest only instantiate encoders using encoder factories.
> * Add HW encoder factories to VideoQualityTest.
> * Add full stack tests:
> - sqcif7 at 30 kbps: libvpx.
> - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
>
> BUG=webrtc:8219
> TBR=asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
>
> Change-Id: I464409ac0d5276defa78c1bf66034c6cca717d74
> Reviewed-on: https://webrtc-review.googlesource.com/4740
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20041}
TBR=kjellander@webrtc.org,brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
Change-Id: If558b7fb86740658e50a6897d1eeeb72103a54ec
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8219
Reviewed-on: https://webrtc-review.googlesource.com/4900
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20044}
249 lines
8.6 KiB
C++
249 lines
8.6 KiB
C++
/*
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* Copyright (c) 2014 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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#ifndef TEST_CALL_TEST_H_
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#define TEST_CALL_TEST_H_
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#include <memory>
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#include <vector>
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#include "call/call.h"
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#include "call/rtp_transport_controller_send.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "test/encoder_settings.h"
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#include "test/fake_audio_device.h"
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#include "test/fake_decoder.h"
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#include "test/fake_encoder.h"
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#include "test/fake_videorenderer.h"
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#include "test/frame_generator_capturer.h"
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#include "test/rtp_rtcp_observer.h"
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#include "test/single_threaded_task_queue.h"
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namespace webrtc {
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class VoEBase;
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namespace test {
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class BaseTest;
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class CallTest : public ::testing::Test {
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public:
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CallTest();
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virtual ~CallTest();
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static const size_t kNumSsrcs = 3;
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static const int kDefaultWidth = 320;
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static const int kDefaultHeight = 180;
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static const int kDefaultFramerate = 30;
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static const int kDefaultTimeoutMs;
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static const int kLongTimeoutMs;
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static const uint8_t kVideoSendPayloadType;
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static const uint8_t kSendRtxPayloadType;
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static const uint8_t kFakeVideoSendPayloadType;
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static const uint8_t kRedPayloadType;
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static const uint8_t kRtxRedPayloadType;
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static const uint8_t kUlpfecPayloadType;
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static const uint8_t kFlexfecPayloadType;
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static const uint8_t kAudioSendPayloadType;
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static const uint8_t kPayloadTypeH264;
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static const uint8_t kPayloadTypeVP8;
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static const uint8_t kPayloadTypeVP9;
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static const uint32_t kSendRtxSsrcs[kNumSsrcs];
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static const uint32_t kVideoSendSsrcs[kNumSsrcs];
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static const uint32_t kAudioSendSsrc;
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static const uint32_t kFlexfecSendSsrc;
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static const uint32_t kReceiverLocalVideoSsrc;
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static const uint32_t kReceiverLocalAudioSsrc;
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static const int kNackRtpHistoryMs;
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static const uint8_t kDefaultKeepalivePayloadType;
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static const std::map<uint8_t, MediaType> payload_type_map_;
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protected:
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// RunBaseTest overwrites the audio_state and the voice_engine of the send and
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// receive Call configs to simplify test code and avoid having old VoiceEngine
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// APIs in the tests.
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void RunBaseTest(BaseTest* test);
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void CreateCalls(const Call::Config& sender_config,
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const Call::Config& receiver_config);
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void CreateSenderCall(const Call::Config& config);
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void CreateReceiverCall(const Call::Config& config);
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void DestroyCalls();
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void CreateSendConfig(size_t num_video_streams,
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size_t num_audio_streams,
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size_t num_flexfec_streams,
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Transport* send_transport);
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void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
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void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
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float speed,
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int framerate,
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int width,
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int height);
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void CreateFrameGeneratorCapturer(int framerate, int width, int height);
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void CreateFakeAudioDevices(
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std::unique_ptr<FakeAudioDevice::Capturer> capturer,
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std::unique_ptr<FakeAudioDevice::Renderer> renderer);
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void CreateVideoStreams();
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void CreateAudioStreams();
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void CreateFlexfecStreams();
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void AssociateFlexfecStreamsWithVideoStreams();
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void DissociateFlexfecStreamsFromVideoStreams();
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void Start();
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void Stop();
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void DestroyStreams();
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void SetFakeVideoCaptureRotation(VideoRotation rotation);
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Clock* const clock_;
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std::unique_ptr<webrtc::RtcEventLog> event_log_;
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std::unique_ptr<Call> sender_call_;
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RtpTransportControllerSend* sender_call_transport_controller_;
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std::unique_ptr<PacketTransport> send_transport_;
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VideoSendStream::Config video_send_config_;
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VideoEncoderConfig video_encoder_config_;
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VideoSendStream* video_send_stream_;
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AudioSendStream::Config audio_send_config_;
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AudioSendStream* audio_send_stream_;
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std::unique_ptr<Call> receiver_call_;
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std::unique_ptr<PacketTransport> receive_transport_;
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std::vector<VideoReceiveStream::Config> video_receive_configs_;
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std::vector<VideoReceiveStream*> video_receive_streams_;
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std::vector<AudioReceiveStream::Config> audio_receive_configs_;
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std::vector<AudioReceiveStream*> audio_receive_streams_;
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std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
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std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
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std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
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test::FakeEncoder fake_encoder_;
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std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_;
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size_t num_video_streams_;
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size_t num_audio_streams_;
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size_t num_flexfec_streams_;
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
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rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
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test::FakeVideoRenderer fake_renderer_;
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SingleThreadedTaskQueueForTesting task_queue_;
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private:
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// TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
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// These methods are used to set up legacy voice engines and channels which is
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// necessary while voice engine is being refactored to the new stream API.
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struct VoiceEngineState {
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VoiceEngineState()
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: voice_engine(nullptr),
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base(nullptr),
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channel_id(-1) {}
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VoiceEngine* voice_engine;
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VoEBase* base;
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int channel_id;
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};
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void CreateVoiceEngines();
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void DestroyVoiceEngines();
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VoiceEngineState voe_send_;
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VoiceEngineState voe_recv_;
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rtc::scoped_refptr<AudioProcessing> apm_send_;
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rtc::scoped_refptr<AudioProcessing> apm_recv_;
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// The audio devices must outlive the voice engines.
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std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_;
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std::unique_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
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};
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class BaseTest : public RtpRtcpObserver {
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public:
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BaseTest();
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explicit BaseTest(unsigned int timeout_ms);
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virtual ~BaseTest();
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virtual void PerformTest() = 0;
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virtual bool ShouldCreateReceivers() const = 0;
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virtual size_t GetNumVideoStreams() const;
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virtual size_t GetNumAudioStreams() const;
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virtual size_t GetNumFlexfecStreams() const;
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virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer();
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virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer();
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virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
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FakeAudioDevice* recv_audio_device);
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virtual Call::Config GetSenderCallConfig();
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virtual Call::Config GetReceiverCallConfig();
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virtual void OnRtpTransportControllerSendCreated(
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RtpTransportControllerSend* controller);
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virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
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virtual test::PacketTransport* CreateSendTransport(
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SingleThreadedTaskQueueForTesting* task_queue,
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Call* sender_call);
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virtual test::PacketTransport* CreateReceiveTransport(
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SingleThreadedTaskQueueForTesting* task_queue);
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virtual void ModifyVideoConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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VideoEncoderConfig* encoder_config);
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virtual void ModifyVideoCaptureStartResolution(int* width,
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int* heigt,
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int* frame_rate);
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virtual void OnVideoStreamsCreated(
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VideoSendStream* send_stream,
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const std::vector<VideoReceiveStream*>& receive_streams);
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virtual void ModifyAudioConfigs(
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AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStream::Config>* receive_configs);
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virtual void OnAudioStreamsCreated(
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AudioSendStream* send_stream,
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const std::vector<AudioReceiveStream*>& receive_streams);
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virtual void ModifyFlexfecConfigs(
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std::vector<FlexfecReceiveStream::Config>* receive_configs);
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virtual void OnFlexfecStreamsCreated(
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const std::vector<FlexfecReceiveStream*>& receive_streams);
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virtual void OnFrameGeneratorCapturerCreated(
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FrameGeneratorCapturer* frame_generator_capturer);
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virtual void OnStreamsStopped();
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std::unique_ptr<webrtc::RtcEventLog> event_log_;
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};
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class SendTest : public BaseTest {
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public:
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explicit SendTest(unsigned int timeout_ms);
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bool ShouldCreateReceivers() const override;
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};
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class EndToEndTest : public BaseTest {
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public:
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EndToEndTest();
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explicit EndToEndTest(unsigned int timeout_ms);
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bool ShouldCreateReceivers() const override;
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};
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} // namespace test
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} // namespace webrtc
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#endif // TEST_CALL_TEST_H_
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