Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/mediastreamprovider.h

+/*
+ *  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_