rhubarb-lip-sync/lib/webrtc-8d2248ff/webrtc/api/mediastreamprovider.h

121 lines
4.4 KiB
C++

/*
* Copyright 2012 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 WEBRTC_API_MEDIASTREAMPROVIDER_H_
#define WEBRTC_API_MEDIASTREAMPROVIDER_H_
#include <memory>
#include "webrtc/api/rtpsenderinterface.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h"
namespace cricket {
class AudioSource;
class VideoFrame;
struct AudioOptions;
struct VideoOptions;
} // namespace cricket
namespace webrtc {
class AudioSinkInterface;
// TODO(deadbeef): Change the key from an ssrc to a "sender_id" or
// "receiver_id" string, which will be the MSID in the short term and MID in
// the long term.
// TODO(deadbeef): These interfaces are effectively just a way for the
// RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
// refactored away eventually, as the classes converge.
// This interface is called by AudioRtpSender/Receivers to change the settings
// of an audio track connected to certain PeerConnection.
class AudioProviderInterface {
public:
// Enable/disable the audio playout of a remote audio track with |ssrc|.
virtual void SetAudioPlayout(uint32_t ssrc, bool enable) = 0;
// Enable/disable sending audio on the local audio track with |ssrc|.
// When |enable| is true |options| should be applied to the audio track.
virtual void SetAudioSend(uint32_t ssrc,
bool enable,
const cricket::AudioOptions& options,
cricket::AudioSource* source) = 0;
// Sets the audio playout volume of a remote audio track with |ssrc|.
// |volume| is in the range of [0, 10].
virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0;
// Allows for setting a direct audio sink for an incoming audio source.
// Only one audio sink is supported per ssrc and ownership of the sink is
// passed to the provider.
virtual void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
virtual RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const = 0;
virtual bool SetAudioRtpSendParameters(uint32_t ssrc,
const RtpParameters& parameters) = 0;
virtual RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const = 0;
virtual bool SetAudioRtpReceiveParameters(
uint32_t ssrc,
const RtpParameters& parameters) = 0;
// Called when the first audio packet is received.
sigslot::signal0<> SignalFirstAudioPacketReceived;
protected:
virtual ~AudioProviderInterface() {}
};
// This interface is called by VideoRtpSender/Receivers to change the settings
// of a video track connected to a certain PeerConnection.
class VideoProviderInterface {
public:
// Enable/disable the video playout of a remote video track with |ssrc|.
virtual void SetVideoPlayout(
uint32_t ssrc,
bool enable,
rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0;
// Enable/disable sending video on the local video track with |ssrc|.
// TODO(deadbeef): Make |options| a reference parameter.
// TODO(deadbeef): Eventually, |enable| and |options| will be contained
// in |source|. When that happens, remove those parameters and rename
// this to SetVideoSource.
virtual void SetVideoSend(
uint32_t ssrc,
bool enable,
const cricket::VideoOptions* options,
rtc::VideoSourceInterface<cricket::VideoFrame>* source) = 0;
virtual RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const = 0;
virtual bool SetVideoRtpSendParameters(uint32_t ssrc,
const RtpParameters& parameters) = 0;
virtual RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const = 0;
virtual bool SetVideoRtpReceiveParameters(
uint32_t ssrc,
const RtpParameters& parameters) = 0;
// Called when the first video packet is received.
sigslot::signal0<> SignalFirstVideoPacketReceived;
protected:
virtual ~VideoProviderInterface() {}
};
} // namespace webrtc
#endif // WEBRTC_API_MEDIASTREAMPROVIDER_H_