blob: 9d182187e3cdd620ad0980bacbc28658470d78ae [file] [log] [blame]
* 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.
#include <memory>
#include <vector>
#include "api/media_transport_interface.h"
#include "api/task_queue/task_queue_factory.h"
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "call/video_receive_stream.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/synchronization/sequence_checker.h"
#include "rtc_base/task_queue.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 ProcessThread;
class RTPFragmentationHeader;
class RtpStreamReceiverInterface;
class RtpStreamReceiverControllerInterface;
class RtxReceiveStream;
class VCMTiming;
namespace internal {
class VideoReceiveStream : public webrtc::VideoReceiveStream,
public rtc::VideoSinkInterface<VideoFrame>,
public NackSender,
public video_coding::OnCompleteFrameCallback,
public Syncable,
public CallStatsObserver,
public MediaTransportVideoSinkInterface,
public MediaTransportRttObserver {
VideoReceiveStream(TaskQueueFactory* task_queue_factory,
RtpStreamReceiverControllerInterface* receiver_controller,
int num_cpu_cores,
PacketRouter* packet_router,
VideoReceiveStream::Config config,
ProcessThread* process_thread,
CallStats* call_stats,
Clock* clock,
VCMTiming* timing);
VideoReceiveStream(TaskQueueFactory* task_queue_factory,
RtpStreamReceiverControllerInterface* receiver_controller,
int num_cpu_cores,
PacketRouter* packet_router,
VideoReceiveStream::Config config,
ProcessThread* process_thread,
CallStats* call_stats,
Clock* clock);
~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;
void AddSecondarySink(RtpPacketSinkInterface* sink) override;
void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override;
// SetBaseMinimumPlayoutDelayMs and GetBaseMinimumPlayoutDelayMs are called
// from webrtc/api level and requested by user code. For e.g. blink/js layer
// in Chromium.
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
int GetBaseMinimumPlayoutDelayMs() const override;
void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
// Implements rtc::VideoSinkInterface<VideoFrame>.
void OnFrame(const VideoFrame& video_frame) override;
// Implements NackSender.
void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
// For this particular override of the interface,
// only (buffering_allowed == true) is acceptable.
void SendNack(const std::vector<uint16_t>& sequence_numbers,
bool buffering_allowed) override;
// Implements video_coding::OnCompleteFrameCallback.
void OnCompleteFrame(
std::unique_ptr<video_coding::EncodedFrame> frame) override;
// Implements MediaTransportVideoSinkInterface, converts the received frame to
// OnCompleteFrameCallback
void OnData(uint64_t channel_id,
MediaTransportEncodedVideoFrame frame) override;
// Implements CallStatsObserver::OnRttUpdate
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
// Implements MediaTransportRttObserver::OnRttUpdated
void OnRttUpdated(int64_t rtt_ms) override;
// Implements Syncable.
int id() const override;
absl::optional<Syncable::Info> GetInfo() const override;
uint32_t GetPlayoutTimestamp() const override;
// SetMinimumPlayoutDelay is only called by A/V sync.
void SetMinimumPlayoutDelay(int delay_ms) override;
std::vector<webrtc::RtpSource> GetSources() const override;
int64_t GetWaitMs() const;
void StartNextDecode() RTC_RUN_ON(decode_queue_);
static void DecodeThreadFunction(void* ptr);
bool Decode();
void HandleEncodedFrame(std::unique_ptr<video_coding::EncodedFrame> frame);
void HandleFrameBufferTimeout();
void UpdatePlayoutDelays() const
void RequestKeyFrame();
SequenceChecker worker_sequence_checker_;
SequenceChecker module_process_sequence_checker_;
SequenceChecker network_sequence_checker_;
TaskQueueFactory* const task_queue_factory_;
TransportAdapter transport_adapter_;
const VideoReceiveStream::Config config_;
const int num_cpu_cores_;
ProcessThread* const process_thread_;
Clock* const clock_;
const bool use_task_queue_;
rtc::PlatformThread decode_thread_;
CallStats* const call_stats_;
bool decoder_running_ RTC_GUARDED_BY(worker_sequence_checker_) = false;
bool decoder_stopped_ RTC_GUARDED_BY(decode_queue_) = true;
ReceiveStatisticsProxy stats_proxy_;
// 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_;
RtpVideoStreamReceiver rtp_video_stream_receiver_;
std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
RtpStreamsSynchronizer rtp_stream_sync_;
// TODO(nisse, philipel): Creation and ownership of video encoders should be
// moved to the new VideoStreamDecoder.
std::vector<std::unique_ptr<VideoDecoder>> video_decoders_;
// Members for the new jitter buffer experiment.
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;
int64_t last_complete_frame_time_ms_ = 0;
// Keyframe request intervals are configurable through field trials.
const int max_wait_for_keyframe_ms_;
const int max_wait_for_frame_ms_;
rtc::CriticalSection playout_delay_lock_;
// All of them tries to change current min_playout_delay on |timing_| but
// source of the change request is different in each case. Among them the
// biggest delay is used. -1 means use default value from the |timing_|.
// Minimum delay as decided by the RTP playout delay extension.
int frame_minimum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;
// Minimum delay as decided by the setLatency function in "webrtc/api".
int base_minimum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;
// Minimum delay as decided by the A/V synchronization feature.
int syncable_minimum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) =
// Maximum delay as decided by the RTP playout delay extension.
int frame_maximum_playout_delay_ms_ RTC_GUARDED_BY(playout_delay_lock_) = -1;
// Defined last so they are destroyed before all other members.
rtc::TaskQueue decode_queue_;
} // namespace internal
} // namespace webrtc