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