Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/mediastreamprovider.h

Index: webrtc/api/mediastreamprovider.h
diff --git a/webrtc/api/mediastreamprovider.h b/webrtc/api/mediastreamprovider.h
deleted file mode 100644
index 784a95423dc9afe423d90c94184b70e9c7de4161..0000000000000000000000000000000000000000
--- a/webrtc/api/mediastreamprovider.h
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- *  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_