141 lines
4.8 KiB
C++
141 lines
4.8 KiB
C++
/*
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* Copyright 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_AUDIO_RTP_RECEIVER_H_
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#define PC_AUDIO_RTP_RECEIVER_H_
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#include <stdint.h>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/crypto/frame_decryptor_interface.h"
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#include "api/media_stream_interface.h"
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#include "api/media_types.h"
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#include "api/rtp_parameters.h"
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#include "api/scoped_refptr.h"
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#include "media/base/media_channel.h"
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#include "pc/jitter_buffer_delay_interface.h"
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#include "pc/remote_audio_source.h"
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#include "pc/rtp_receiver.h"
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#include "rtc_base/ref_counted_object.h"
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#include "rtc_base/thread.h"
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namespace webrtc {
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class AudioRtpReceiver : public ObserverInterface,
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public AudioSourceInterface::AudioObserver,
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public rtc::RefCountedObject<RtpReceiverInternal> {
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public:
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AudioRtpReceiver(rtc::Thread* worker_thread,
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std::string receiver_id,
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std::vector<std::string> stream_ids);
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// TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
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AudioRtpReceiver(
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rtc::Thread* worker_thread,
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const std::string& receiver_id,
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const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
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virtual ~AudioRtpReceiver();
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// ObserverInterface implementation
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void OnChanged() override;
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// AudioSourceInterface::AudioObserver implementation
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void OnSetVolume(double volume) override;
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rtc::scoped_refptr<AudioTrackInterface> audio_track() const {
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return track_.get();
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}
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// RtpReceiverInterface implementation
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rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
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return track_.get();
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}
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
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return dtls_transport_;
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}
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std::vector<std::string> stream_ids() const override;
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std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
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const override {
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return streams_;
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}
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cricket::MediaType media_type() const override {
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return cricket::MEDIA_TYPE_AUDIO;
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}
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std::string id() const override { return id_; }
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RtpParameters GetParameters() const override;
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void SetFrameDecryptor(
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
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rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
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const override;
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// RtpReceiverInternal implementation.
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void Stop() override;
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void SetupMediaChannel(uint32_t ssrc) override;
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void SetupUnsignaledMediaChannel() override;
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uint32_t ssrc() const override { return ssrc_.value_or(0); }
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void NotifyFirstPacketReceived() override;
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void set_stream_ids(std::vector<std::string> stream_ids) override;
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void set_transport(
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
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dtls_transport_ = dtls_transport;
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}
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void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
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streams) override;
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void SetObserver(RtpReceiverObserverInterface* observer) override;
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void SetJitterBufferMinimumDelay(
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absl::optional<double> delay_seconds) override;
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void SetMediaChannel(cricket::MediaChannel* media_channel) override;
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std::vector<RtpSource> GetSources() const override;
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int AttachmentId() const override { return attachment_id_; }
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void SetDepacketizerToDecoderFrameTransformer(
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rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
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override;
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private:
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void RestartMediaChannel(absl::optional<uint32_t> ssrc);
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void Reconfigure();
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bool SetOutputVolume(double volume);
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rtc::Thread* const worker_thread_;
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const std::string id_;
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const rtc::scoped_refptr<RemoteAudioSource> source_;
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const rtc::scoped_refptr<AudioTrackInterface> track_;
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cricket::VoiceMediaChannel* media_channel_ = nullptr;
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absl::optional<uint32_t> ssrc_;
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std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
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bool cached_track_enabled_;
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double cached_volume_ = 1;
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bool stopped_ = true;
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RtpReceiverObserverInterface* observer_ = nullptr;
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bool received_first_packet_ = false;
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int attachment_id_ = 0;
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
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rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
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// Allows to thread safely change playout delay. Handles caching cases if
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// |SetJitterBufferMinimumDelay| is called before start.
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rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
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RTC_GUARDED_BY(worker_thread_);
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};
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} // namespace webrtc
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#endif // PC_AUDIO_RTP_RECEIVER_H_
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