webrtc_m130/pc/audio_rtp_receiver.h
Tommi 20d8d9150c Reland "Remove stopped_ from AudioRtpReceiver and VideoRtpReceiver."
This is a reland of 3ed36c0521546881656c73984456485dcab16205

Original change's description:
> Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver."
>
> This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f
>
> The difference from the original CL is that a check for
> `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
> This caused a side effect that registering the sink while the source
> was in an "initializing" state, failed. The last remaining state
> however, is `kEnded` - but since there's no logic in the class around
> the expected value of the states, the check inside of AddSink()
> doesn't provide an additional value - it's rather a surprise for
> developers if it doesn't succeed. So, now removed.
>
> Original change's description:
> > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
> >
> > This simplifies the logic in these classes a bit, which makes upcoming
> > change easier. The `stopped_` flag in these classes was essentially
> > the same thing as `media_channel_ == nullptr`, which is what's
> > consistently used now for the same checks.
> >
> > Bug: webrtc:13540
> > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35907}
>
> Bug: webrtc:13540
> Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35958}

Bug: webrtc:13540
Change-Id: I6d7d67fddb1ddfc69a302f0f69a9b815f2fd82f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251386
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35967}
2022-02-09 16:54:06 +00:00

156 lines
5.9 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_AUDIO_RTP_RECEIVER_H_
#define PC_AUDIO_RTP_RECEIVER_H_
#include <stdint.h>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/dtls_transport_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/sequence_checker.h"
#include "api/transport/rtp/rtp_source.h"
#include "media/base/media_channel.h"
#include "pc/audio_track.h"
#include "pc/jitter_buffer_delay.h"
#include "pc/media_stream_track_proxy.h"
#include "pc/remote_audio_source.h"
#include "pc/rtp_receiver.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class AudioRtpReceiver : public ObserverInterface,
public AudioSourceInterface::AudioObserver,
public RtpReceiverInternal {
public:
AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids,
bool is_unified_plan);
// TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
AudioRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
bool is_unified_plan);
virtual ~AudioRtpReceiver();
// ObserverInterface implementation
void OnChanged() override;
// AudioSourceInterface::AudioObserver implementation
void OnSetVolume(double volume) override;
rtc::scoped_refptr<AudioTrackInterface> audio_track() const { return track_; }
// RtpReceiverInterface implementation
rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_;
}
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override;
std::vector<std::string> stream_ids() const override;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
const override;
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
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;
// RtpReceiverInternal implementation.
void Stop() override;
void StopAndEndTrack() override;
void SetupMediaChannel(uint32_t ssrc) override;
void SetupUnsignaledMediaChannel() override;
uint32_t ssrc() const override;
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override;
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;
std::vector<RtpSource> GetSources() const override;
int AttachmentId() const override { return attachment_id_; }
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
private:
void RestartMediaChannel(absl::optional<uint32_t> ssrc);
void Reconfigure(bool track_enabled, double volume)
RTC_RUN_ON(worker_thread_);
void SetOutputVolume_w(double volume) RTC_RUN_ON(worker_thread_);
void SetMediaChannel_w(cricket::MediaChannel* media_channel)
RTC_RUN_ON(worker_thread_);
RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
rtc::Thread* const worker_thread_;
const std::string id_;
const rtc::scoped_refptr<RemoteAudioSource> source_;
const rtc::scoped_refptr<AudioTrackProxyWithInternal<AudioTrack>> track_;
cricket::VoiceMediaChannel* media_channel_ RTC_GUARDED_BY(worker_thread_) =
nullptr;
absl::optional<uint32_t> ssrc_ RTC_GUARDED_BY(worker_thread_);
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_
RTC_GUARDED_BY(&signaling_thread_checker_);
bool cached_track_enabled_ RTC_GUARDED_BY(&signaling_thread_checker_);
double cached_volume_ RTC_GUARDED_BY(&signaling_thread_checker_) = 1.0;
RtpReceiverObserverInterface* observer_
RTC_GUARDED_BY(&signaling_thread_checker_) = nullptr;
bool received_first_packet_ RTC_GUARDED_BY(&signaling_thread_checker_) =
false;
const int attachment_id_;
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_
RTC_GUARDED_BY(worker_thread_);
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_
RTC_GUARDED_BY(&signaling_thread_checker_);
// Stores and updates the playout delay. Handles caching cases if
// `SetJitterBufferMinimumDelay` is called before start.
JitterBufferDelay delay_ RTC_GUARDED_BY(worker_thread_);
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
RTC_GUARDED_BY(worker_thread_);
const rtc::scoped_refptr<PendingTaskSafetyFlag> worker_thread_safety_;
};
} // namespace webrtc
#endif // PC_AUDIO_RTP_RECEIVER_H_