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/*
* 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.
*/
#ifndef VIDEO_VIDEO_RECEIVE_STREAM2_H_
#define VIDEO_VIDEO_RECEIVE_STREAM2_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.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/source/source_tracker.h"
#include "modules/video_coding/nack_requester.h"
#include "modules/video_coding/video_receiver2.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"
#include "video/receive_statistics_proxy.h"
#include "video/rtp_streams_synchronizer2.h"
#include "video/rtp_video_stream_receiver2.h"
#include "video/transport_adapter.h"
#include "video/video_stream_buffer_controller.h"
#include "video/video_stream_decoder2.h"
namespace webrtc {
class RtpStreamReceiverInterface;
class RtpStreamReceiverControllerInterface;
class RtxReceiveStream;
class VCMTiming;
constexpr TimeDelta kMaxWaitForKeyFrame = TimeDelta::Millis(200);
constexpr TimeDelta kMaxWaitForFrame = TimeDelta::Seconds(3);
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 needing
// multiple calls to clock->Now().
struct VideoFrameMetaData {
VideoFrameMetaData(const webrtc::VideoFrame& frame, Timestamp now)
: rtp_timestamp(frame.rtp_timestamp()),
timestamp_us(frame.timestamp_us()),
ntp_time_ms(frame.ntp_time_ms()),
width(frame.width()),
height(frame.height()),
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::VideoReceiveStreamInterface,
public rtc::VideoSinkInterface<VideoFrame>,
public RtpVideoStreamReceiver2::OnCompleteFrameCallback,
public Syncable,
public CallStatsObserver,
public FrameSchedulingReceiver {
public:
// The maximum number of buffered encoded frames when encoded output is
// configured.
static constexpr size_t kBufferedEncodedFramesMaxSize = 60;
VideoReceiveStream2(const Environment& env,
Call* call,
int num_cpu_cores,
PacketRouter* packet_router,
VideoReceiveStreamInterface::Config config,
CallStats* call_stats,
std::unique_ptr<VCMTiming> timing,
NackPeriodicProcessor* nack_periodic_processor,
DecodeSynchronizer* decode_sync);
// 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();
// Accessor for the a/v sync group. This value may change and the caller
// must be on the packet delivery thread.
const std::string& sync_group() const;
// Getters for const remote SSRC values that won't change throughout the
// object's lifetime.
uint32_t remote_ssrc() const { return config_.rtp.remote_ssrc; }
// RTX ssrc can be updated.
uint32_t rtx_ssrc() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return updated_rtx_ssrc_.value_or(config_.rtp.rtx_ssrc);
}
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
void SetSync(Syncable* audio_syncable);
// Updates the `rtp_video_stream_receiver_`'s `local_ssrc` when the default
// sender has been created, changed or removed.
void SetLocalSsrc(uint32_t local_ssrc);
// Implements webrtc::VideoReceiveStreamInterface.
void Start() override;
void Stop() override;
void SetRtcpMode(RtcpMode mode) override;
void SetFlexFecProtection(RtpPacketSinkInterface* flexfec_sink) override;
void SetLossNotificationEnabled(bool enabled) override;
void SetNackHistory(TimeDelta history) override;
void SetProtectionPayloadTypes(int red_payload_type,
int ulpfec_payload_type) override;
void SetRtcpXr(Config::Rtp::RtcpXr rtcp_xr) override;
void SetAssociatedPayloadTypes(
std::map<int, int> associated_payload_types) override;
webrtc::VideoReceiveStreamInterface::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 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 UpdateRtxSsrc(uint32_t ssrc) override;
private:
// FrameSchedulingReceiver implementation.
// Called on packet sequence.
void OnEncodedFrame(std::unique_ptr<EncodedFrame> frame) override;
// Called on packet sequence.
void OnDecodableFrameTimeout(TimeDelta wait) override;
void CreateAndRegisterExternalDecoder(const Decoder& decoder);
struct DecodeFrameResult {
// True if the decoder returned code WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME,
// or if the decoder failed and a keyframe is required. When true, a
// keyframe request should be sent even if a keyframe request was sent
// recently.
bool force_request_key_frame;
// The picture id of the frame that was decoded, or nullopt if the frame was
// not decoded.
absl::optional<int64_t> decoded_frame_picture_id;
// True if the next frame decoded must be a keyframe. This value will set
// the value of `keyframe_required_`, which will force the frame buffer to
// drop all frames that are not keyframes.
bool keyframe_required;
};
DecodeFrameResult HandleEncodedFrameOnDecodeQueue(
std::unique_ptr<EncodedFrame> frame,
bool keyframe_request_is_due,
bool keyframe_required) RTC_RUN_ON(decode_sequence_checker_);
void UpdatePlayoutDelays() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_sequence_checker_);
void RequestKeyFrame(Timestamp now) RTC_RUN_ON(packet_sequence_checker_);
void HandleKeyFrameGeneration(bool received_frame_is_keyframe,
Timestamp now,
bool always_request_key_frame,
bool keyframe_request_is_due)
RTC_RUN_ON(packet_sequence_checker_);
bool IsReceivingKeyFrame(Timestamp timestamp) const
RTC_RUN_ON(packet_sequence_checker_);
int DecodeAndMaybeDispatchEncodedFrame(std::unique_ptr<EncodedFrame> frame)
RTC_RUN_ON(decode_sequence_checker_);
void UpdateHistograms();
const Environment env_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_sequence_checker_;
// TODO(bugs.webrtc.org/11993): 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_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker decode_sequence_checker_;
TransportAdapter transport_adapter_;
const VideoReceiveStreamInterface::Config config_;
const int num_cpu_cores_;
Call* const call_;
CallStats* const call_stats_;
bool decoder_running_ RTC_GUARDED_BY(worker_sequence_checker_) = false;
bool decoder_stopped_ RTC_GUARDED_BY(decode_sequence_checker_) = 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_;
std::unique_ptr<VideoStreamBufferController> buffer_;
// `receiver_controller_` is valid from when RegisterWithTransport is invoked
// until UnregisterFromTransport.
RtpStreamReceiverControllerInterface* receiver_controller_
RTC_GUARDED_BY(packet_sequence_checker_) = nullptr;
std::unique_ptr<RtpStreamReceiverInterface> media_receiver_
RTC_GUARDED_BY(packet_sequence_checker_);
std::unique_ptr<RtxReceiveStream> rtx_receive_stream_
RTC_GUARDED_BY(packet_sequence_checker_);
absl::optional<uint32_t> updated_rtx_ssrc_
RTC_GUARDED_BY(packet_sequence_checker_);
std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_
RTC_GUARDED_BY(packet_sequence_checker_);
// 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(packet_sequence_checker_) = true;
// If we have successfully decoded any frame.
bool frame_decoded_ RTC_GUARDED_BY(decode_sequence_checker_) = false;
absl::optional<Timestamp> last_keyframe_request_
RTC_GUARDED_BY(packet_sequence_checker_);
// Keyframe request intervals are configurable through field trials.
TimeDelta max_wait_for_keyframe_ RTC_GUARDED_BY(packet_sequence_checker_);
TimeDelta max_wait_for_frame_ RTC_GUARDED_BY(packet_sequence_checker_);
// 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.
absl::optional<TimeDelta> frame_minimum_playout_delay_
RTC_GUARDED_BY(worker_sequence_checker_);
// Minimum delay as decided by the setLatency function in "webrtc/api".
absl::optional<TimeDelta> base_minimum_playout_delay_
RTC_GUARDED_BY(worker_sequence_checker_);
// Minimum delay as decided by the A/V synchronization feature.
absl::optional<TimeDelta> syncable_minimum_playout_delay_
RTC_GUARDED_BY(worker_sequence_checker_);
// Maximum delay as decided by the RTP playout delay extension.
absl::optional<TimeDelta> frame_maximum_playout_delay_
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_sequence_checker_);
// Set to true while we're requesting keyframes but not yet received one.
bool keyframe_generation_requested_ RTC_GUARDED_BY(packet_sequence_checker_) =
false;
// 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_
RTC_GUARDED_BY(pending_resolution_mutex_);
// Buffered encoded frames held while waiting for decoded resolution.
std::vector<std::unique_ptr<EncodedFrame>> buffered_encoded_frames_
RTC_GUARDED_BY(decode_sequence_checker_);
// Used to signal destruction to potentially pending tasks.
ScopedTaskSafety task_safety_;
// Defined last so they are destroyed before all other members, in particular
// `decode_queue_` should be stopped before `decode_sequence_checker_` is
// destructed to avoid races when running tasks on the `decode_queue_` during
// VideoReceiveStream2 destruction.
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> decode_queue_;
absl::optional<uint32_t> last_decoded_rtp_timestamp_;
};
} // namespace internal
} // namespace webrtc
#endif // VIDEO_VIDEO_RECEIVE_STREAM2_H_