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/*
* Copyright (c) 2013 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.
*/
#ifndef VIDEO_VIDEO_RECEIVE_STREAM_H_
#define VIDEO_VIDEO_RECEIVE_STREAM_H_
#include <memory>
#include <vector>
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "call/video_receive_stream.h"
#include "common_video/include/incoming_video_stream.h"
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "modules/rtp_rtcp/include/flexfec_receiver.h"
#include "modules/video_coding/frame_buffer2.h"
#include "modules/video_coding/video_coding_impl.h"
#include "rtc_base/sequenced_task_checker.h"
#include "system_wrappers/include/clock.h"
#include "video/receive_statistics_proxy.h"
#include "video/rtp_streams_synchronizer.h"
#include "video/rtp_video_stream_receiver.h"
#include "video/transport_adapter.h"
#include "video/video_stream_decoder.h"
namespace webrtc {
class CallStats;
class IvfFileWriter;
class ProcessThread;
class RTPFragmentationHeader;
class RtpStreamReceiverInterface;
class RtpStreamReceiverControllerInterface;
class RtxReceiveStream;
class VCMTiming;
class VCMJitterEstimator;
namespace internal {
class VideoReceiveStream : public webrtc::VideoReceiveStream,
public rtc::VideoSinkInterface<VideoFrame>,
public EncodedImageCallback,
public NackSender,
public KeyFrameRequestSender,
public video_coding::OnCompleteFrameCallback,
public Syncable,
public CallStatsObserver {
public:
VideoReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
int num_cpu_cores,
PacketRouter* packet_router,
VideoReceiveStream::Config config,
ProcessThread* process_thread,
CallStats* call_stats);
~VideoReceiveStream() override;
const Config& config() const { return config_; }
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
void SetSync(Syncable* audio_syncable);
// Implements webrtc::VideoReceiveStream.
void Start() override;
void Stop() override;
webrtc::VideoReceiveStream::Stats GetStats() const override;
// Takes ownership of the file, is responsible for closing it later.
// Calling this method will close and finalize any current log.
// Giving rtc::kInvalidPlatformFileValue disables logging.
// If a frame to be written would make the log too large the write fails and
// the log is closed and finalized. A |byte_limit| of 0 means no limit.
void EnableEncodedFrameRecording(rtc::PlatformFile file,
size_t byte_limit) override;
void AddSecondarySink(RtpPacketSinkInterface* sink) override;
void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override;
// Implements rtc::VideoSinkInterface<VideoFrame>.
void OnFrame(const VideoFrame& video_frame) override;
// Implements EncodedImageCallback.
EncodedImageCallback::Result OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation) override;
// Implements NackSender.
void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
// Implements KeyFrameRequestSender.
void RequestKeyFrame() override;
// Implements video_coding::OnCompleteFrameCallback.
void OnCompleteFrame(
std::unique_ptr<video_coding::FrameObject> frame) override;
// Implements CallStatsObserver::OnRttUpdate
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
// Implements Syncable.
int id() const override;
rtc::Optional<Syncable::Info> GetInfo() const override;
uint32_t GetPlayoutTimestamp() const override;
void SetMinimumPlayoutDelay(int delay_ms) override;
private:
static void DecodeThreadFunction(void* ptr);
bool Decode();
rtc::SequencedTaskChecker worker_sequence_checker_;
rtc::SequencedTaskChecker module_process_sequence_checker_;
TransportAdapter transport_adapter_;
const VideoReceiveStream::Config config_;
const int num_cpu_cores_;
ProcessThread* const process_thread_;
Clock* const clock_;
rtc::PlatformThread decode_thread_;
CallStats* const call_stats_;
// Shared by media and rtx stream receivers, since the latter has no RtpRtcp
// module of its own.
const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<VCMTiming> timing_; // Jitter buffer experiment.
vcm::VideoReceiver video_receiver_;
std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
ReceiveStatisticsProxy stats_proxy_;
RtpVideoStreamReceiver rtp_video_stream_receiver_;
std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
RtpStreamsSynchronizer rtp_stream_sync_;
rtc::CriticalSection ivf_writer_lock_;
std::unique_ptr<IvfFileWriter> ivf_writer_ RTC_GUARDED_BY(ivf_writer_lock_);
// Members for the new jitter buffer experiment.
std::unique_ptr<VCMJitterEstimator> jitter_estimator_;
std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
std::unique_ptr<RtxReceiveStream> rtx_receive_stream_;
std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
// Whenever we are in an undecodable state (stream has just started or due to
// a decoding error) we require a keyframe to restart the stream.
bool keyframe_required_ = true;
// If we have successfully decoded any frame.
bool frame_decoded_ = false;
int64_t last_keyframe_request_ms_ = 0;
};
} // namespace internal
} // namespace webrtc
#endif // VIDEO_VIDEO_RECEIVE_STREAM_H_
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