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* VoEBase contains only stub methods (until downstream code is updated). * voe::Channel and ChannelProxy classes remain, but are now created internally to the streams. As a result, internal::Audio[Receive|Send]Stream can have a ChannelProxy injected for testing. * Stream classes share Call::module_process_thread_ for their RtpRtcp modules, rather than using a separate thread shared only among audio streams. * voe::Channel instances use Call::worker_queue_ for encoding packets, rather than having a separate queue for audio (send) streams. Bug: webrtc:4690 Change-Id: I8059ef224ad13aa0a6ded2cafc52599c7f64d68d Reviewed-on: https://webrtc-review.googlesource.com/34640 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21578}
236 lines
8.6 KiB
C++
236 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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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 constexpr size_t kNumSsrcs = 6;
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static const int kNumSimulcastStreams = 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 of the send and receive Call configs
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// to simplify test code.
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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 CreateVideoSendConfig(VideoSendStream::Config* video_config,
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size_t num_video_streams,
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size_t num_used_ssrcs,
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Transport* send_transport);
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void CreateAudioAndFecSendConfigs(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 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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std::vector<VideoReceiveStream::Config> CreateMatchingVideoReceiveConfigs(
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const VideoSendStream::Config& video_send_config,
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Transport* rtcp_send_transport);
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void CreateMatchingAudioAndFecConfigs(Transport* rtcp_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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rtc::scoped_refptr<AudioProcessing> apm_send_;
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rtc::scoped_refptr<AudioProcessing> apm_recv_;
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rtc::scoped_refptr<test::FakeAudioDevice> fake_send_audio_device_;
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rtc::scoped_refptr<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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