webrtc/pc/video_rtp_receiver.h
Marina Ciocea 412a31bbf8 Insert frame transformer between Depacketizer and Decoder.
Add a new API in RTReceiverInterface, to be called from the browser side
to insert a frame transformer between the Depacketizer and the Decoder.

The frame transformer is passed from RTReceiverInterface through the
library to be eventually set in RtpVideoStreamReceiver, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169130.

This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30654}
2020-03-02 08:33:44 +00:00

154 lines
5.4 KiB
C++

/*
* Copyright 2019 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 PC_VIDEO_RTP_RECEIVER_H_
#define PC_VIDEO_RTP_RECEIVER_H_
#include <stdint.h>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "media/base/media_channel.h"
#include "pc/jitter_buffer_delay_interface.h"
#include "pc/rtp_receiver.h"
#include "pc/video_rtp_track_source.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/thread.h"
namespace webrtc {
class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
public VideoRtpTrackSource::Callback {
public:
// An SSRC of 0 will create a receiver that will match the first SSRC it
// sees. Must be called on signaling thread.
VideoRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> streams_ids);
// TODO(hbos): Remove this when streams() is removed.
// https://crbug.com/webrtc/9480
VideoRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
virtual ~VideoRtpReceiver();
rtc::scoped_refptr<VideoTrackInterface> video_track() const {
return track_.get();
}
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_.get();
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
return dtls_transport_;
}
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override {
return streams_;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
std::string id() const override { return id_; }
RtpParameters GetParameters() const override;
void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
const override;
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;
// RtpReceiverInternal implementation.
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
void SetupUnsignaledMediaChannel() override;
uint32_t ssrc() const override { return ssrc_.value_or(0); }
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
dtls_transport_ = dtls_transport;
}
void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds) override;
void SetMediaChannel(cricket::MediaChannel* media_channel) override;
int AttachmentId() const override { return attachment_id_; }
std::vector<RtpSource> GetSources() const override;
private:
void RestartMediaChannel(absl::optional<uint32_t> ssrc);
void SetSink(rtc::VideoSinkInterface<VideoFrame>* sink)
RTC_RUN_ON(worker_thread_);
// VideoRtpTrackSource::Callback
void OnGenerateKeyFrame() override;
void OnEncodedSinkEnabled(bool enable) override;
void SetEncodedSinkEnabled(bool enable) RTC_RUN_ON(worker_thread_);
rtc::Thread* const worker_thread_;
const std::string id_;
cricket::VideoMediaChannel* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
// |source_| is held here to be able to change the state of the source when
// the VideoRtpReceiver is stopped.
rtc::scoped_refptr<VideoRtpTrackSource> source_;
rtc::scoped_refptr<VideoTrackInterface> track_;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
bool stopped_ = true;
RtpReceiverObserverInterface* observer_ = nullptr;
bool received_first_packet_ = false;
int attachment_id_ = 0;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
RTC_GUARDED_BY(worker_thread_);
// Allows to thread safely change jitter buffer delay. Handles caching cases
// if |SetJitterBufferMinimumDelay| is called before start.
rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
// Records if we should generate a keyframe when |media_channel_| gets set up
// or switched.
bool saved_generate_keyframe_ RTC_GUARDED_BY(worker_thread_) = false;
bool saved_encoded_sink_enabled_ RTC_GUARDED_BY(worker_thread_) = false;
};
} // namespace webrtc
#endif // PC_VIDEO_RTP_RECEIVER_H_