blob: 95570442778267e80e8db2019a9c338d80febc83 [file] [log] [blame]
* Copyright (c) 2020 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/sequence_checker.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/units/timestamp.h"
#include "api/video/recordable_encoded_frame.h"
#include "call/call.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/rtp_rtcp/source/source_tracker.h"
#include "modules/video_coding/frame_buffer2.h"
#include "modules/video_coding/video_receiver2.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
#include "video/receive_statistics_proxy2.h"
#include "video/rtp_streams_synchronizer2.h"
#include "video/rtp_video_stream_receiver2.h"
#include "video/transport_adapter.h"
#include "video/video_stream_decoder2.h"
namespace webrtc {
class RtpStreamReceiverInterface;
class RtpStreamReceiverControllerInterface;
class RtxReceiveStream;
class VCMTiming;
namespace internal {
class CallStats;
// Utility struct for grabbing metadata from a VideoFrame and processing it
// asynchronously without needing the actual frame data.
// Additionally the caller can bundle information from the current clock
// when the metadata is captured, for accurate reporting and not needeing
// multiple calls to clock->Now().
struct VideoFrameMetaData {
VideoFrameMetaData(const webrtc::VideoFrame& frame, Timestamp now)
: rtp_timestamp(frame.timestamp()),
decode_timestamp(now) {}
int64_t render_time_ms() const {
return timestamp_us / rtc::kNumMicrosecsPerMillisec;
const uint32_t rtp_timestamp;
const int64_t timestamp_us;
const int64_t ntp_time_ms;
const int width;
const int height;
const Timestamp decode_timestamp;
class VideoReceiveStream2
: public webrtc::VideoReceiveStream,
public rtc::VideoSinkInterface<VideoFrame>,
public NackSender,
public RtpVideoStreamReceiver2::OnCompleteFrameCallback,
public Syncable,
public CallStatsObserver {
// The default number of milliseconds to pass before re-requesting a key frame
// to be sent.
static constexpr int kMaxWaitForKeyFrameMs = 200;
// The maximum number of buffered encoded frames when encoded output is
// configured.
static constexpr size_t kBufferedEncodedFramesMaxSize = 60;
VideoReceiveStream2(TaskQueueFactory* task_queue_factory,
Call* call,
int num_cpu_cores,
PacketRouter* packet_router,
VideoReceiveStream::Config config,
CallStats* call_stats,
Clock* clock,
VCMTiming* timing);
// Destruction happens on the worker thread. Prior to destruction the caller
// must ensure that a registration with the transport has been cleared. See
// `RegisterWithTransport` for details.
// TODO(tommi): As a further improvement to this, performing the full
// destruction on the network thread could be made the default.
~VideoReceiveStream2() override;
// Called on `packet_sequence_checker_` to register/unregister with the
// network transport.
void RegisterWithTransport(
RtpStreamReceiverControllerInterface* receiver_controller);
// If registration has previously been done (via `RegisterWithTransport`) then
// `UnregisterFromTransport` must be called prior to destruction, on the
// network thread.
void UnregisterFromTransport();
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;
const RtpConfig& rtp_config() const override { return config_.rtp; }
webrtc::VideoReceiveStream::Stats GetStats() const 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;
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;
// Implements rtc::VideoSinkInterface<VideoFrame>.
void OnFrame(const VideoFrame& video_frame) override;
// Implements NackSender.
// 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 RtpVideoStreamReceiver2::OnCompleteFrameCallback.
void OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) override;
// Implements CallStatsObserver::OnRttUpdate
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
// Implements Syncable.
uint32_t id() const override;
absl::optional<Syncable::Info> GetInfo() const override;
bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const override;
void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
int64_t time_ms) override;
// SetMinimumPlayoutDelay is only called by A/V sync.
bool SetMinimumPlayoutDelay(int delay_ms) override;
std::vector<webrtc::RtpSource> GetSources() const override;
RecordingState SetAndGetRecordingState(RecordingState state,
bool generate_key_frame) override;
void GenerateKeyFrame() override;
void CreateAndRegisterExternalDecoder(const Decoder& decoder);
int64_t GetMaxWaitMs() const RTC_RUN_ON(decode_queue_);
void StartNextDecode() RTC_RUN_ON(decode_queue_);
void HandleEncodedFrame(std::unique_ptr<EncodedFrame> frame)
void HandleFrameBufferTimeout(int64_t now_ms, int64_t wait_ms)
void UpdatePlayoutDelays() const
void RequestKeyFrame(int64_t timestamp_ms)
void HandleKeyFrameGeneration(bool received_frame_is_keyframe,
int64_t now_ms,
bool always_request_key_frame,
bool keyframe_request_is_due)
bool IsReceivingKeyFrame(int64_t timestamp_ms) const
int DecodeAndMaybeDispatchEncodedFrame(std::unique_ptr<EncodedFrame> frame)
void UpdateHistograms();
RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_sequence_checker_;
// TODO( This checker conceptually represents
// operations that belong to the network thread. The Call class is currently
// moving towards handling network packets on the network thread and while
// that work is ongoing, this checker may in practice represent the worker
// thread, but still serves as a mechanism of grouping together concepts
// that belong to the network thread. Once the packets are fully delivered
// on the network thread, this comment will be deleted.
RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_;
TaskQueueFactory* const task_queue_factory_;
TransportAdapter transport_adapter_;
const VideoReceiveStream::Config config_;
const int num_cpu_cores_;
Call* const call_;
Clock* const clock_;
CallStats* const call_stats_;
bool decoder_running_ RTC_GUARDED_BY(worker_sequence_checker_) = false;
bool decoder_stopped_ RTC_GUARDED_BY(decode_queue_) = true;
SourceTracker source_tracker_;
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.
VideoReceiver2 video_receiver_;
std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
RtpVideoStreamReceiver2 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_ RTC_GUARDED_BY(decode_queue_) = true;
// If we have successfully decoded any frame.
bool frame_decoded_ RTC_GUARDED_BY(decode_queue_) = false;
int64_t last_keyframe_request_ms_ RTC_GUARDED_BY(decode_queue_) = 0;
int64_t last_complete_frame_time_ms_
RTC_GUARDED_BY(worker_sequence_checker_) = 0;
// Keyframe request intervals are configurable through field trials.
const int max_wait_for_keyframe_ms_;
const int max_wait_for_frame_ms_;
// 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(worker_sequence_checker_) =
// Minimum delay as decided by the setLatency function in "webrtc/api".
int base_minimum_playout_delay_ms_ RTC_GUARDED_BY(worker_sequence_checker_) =
// Minimum delay as decided by the A/V synchronization feature.
int syncable_minimum_playout_delay_ms_
RTC_GUARDED_BY(worker_sequence_checker_) = -1;
// Maximum delay as decided by the RTP playout delay extension.
int frame_maximum_playout_delay_ms_ RTC_GUARDED_BY(worker_sequence_checker_) =
// Function that is triggered with encoded frames, if not empty.
std::function<void(const RecordableEncodedFrame&)>
encoded_frame_buffer_function_ RTC_GUARDED_BY(decode_queue_);
// Set to true while we're requesting keyframes but not yet received one.
bool keyframe_generation_requested_ RTC_GUARDED_BY(packet_sequence_checker_) =
// Lock to avoid unnecessary per-frame idle wakeups in the code.
webrtc::Mutex pending_resolution_mutex_;
// Signal from decode queue to OnFrame callback to fill pending_resolution_.
// absl::nullopt - no resolution needed. 0x0 - next OnFrame to fill with
// received resolution. Not 0x0 - OnFrame has filled a resolution.
absl::optional<RecordableEncodedFrame::EncodedResolution> pending_resolution_
// Buffered encoded frames held while waiting for decoded resolution.
std::vector<std::unique_ptr<EncodedFrame>> buffered_encoded_frames_
// Set by the field trial WebRTC-LowLatencyRenderer. The parameter |enabled|
// determines if the low-latency renderer algorithm should be used for the
// case min playout delay=0 and max playout delay>0.
FieldTrialParameter<bool> low_latency_renderer_enabled_;
// Set by the field trial WebRTC-LowLatencyRenderer. The parameter
// |include_predecode_buffer| determines if the predecode buffer should be
// taken into account when calculating maximum number of frames in composition
// queue.
FieldTrialParameter<bool> low_latency_renderer_include_predecode_buffer_;
// Set by the field trial WebRTC-PreStreamDecoders. The parameter |max|
// determines the maximum number of decoders that are created up front before
// any video frame has been received.
FieldTrialParameter<int> maximum_pre_stream_decoders_;
// Defined last so they are destroyed before all other members.
rtc::TaskQueue decode_queue_;
// Used to signal destruction to potentially pending tasks.
ScopedTaskSafety task_safety_;
} // namespace internal
} // namespace webrtc