Prevent a voice channel from sending data before a renderer is set.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 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.
  */
 
@@ skipping 24 matching lines @@
 class RateLimiter;
 class Timing;
 }
 
 namespace webrtc {
 class AudioSinkInterface;
 }
 
 namespace cricket {
 
-class AudioRenderer;
+class AudioSource;
 class ScreencastId;
 class VideoCapturer;
 class VideoFrame;
 struct RtpHeader;
 struct VideoFormat;
 
 const int kMinRtpHeaderExtensionId = 1;
 const int kMaxRtpHeaderExtensionId = 255;
 const int kScreencastDefaultFps = 5;
 
@@ skipping 385 matching lines @@
   }
 
   const bool enable_dscp_;
   // |network_interface_| can be accessed from the worker_thread and
   // from any MediaEngine threads. This critical section is to protect accessing
   // of network_interface_ object.
   rtc::CriticalSection network_interface_crit_;
   NetworkInterface* network_interface_;
 };
 
-enum SendFlags {
-  SEND_NOTHING,
-  SEND_MICROPHONE
-};
-
 // The stats information is structured as follows:
 // Media are represented by either MediaSenderInfo or MediaReceiverInfo.
 // Media contains a vector of SSRC infos that are exclusively used by this
 // media. (SSRCs shared between media streams can't be represented.)
 
 // Information about an SSRC.
 // This data may be locally recorded, or received in an RTCP SR or RR.
 struct SsrcSenderInfo {
   SsrcSenderInfo()
       : ssrc(0),
@@ skipping 428 matching lines @@
   };
 
   VoiceMediaChannel() {}
   VoiceMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
   virtual ~VoiceMediaChannel() {}
   virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
   virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
   // Starts or stops playout of received audio.
   virtual bool SetPlayout(bool playout) = 0;
   // Starts or stops sending (and potentially capture) of local audio.
-  virtual bool SetSend(SendFlags flag) = 0;
+  virtual void SetSend(bool send) = 0;
   // Configure stream for sending.
   virtual bool SetAudioSend(uint32_t ssrc,
                             bool enable,
                             const AudioOptions* options,
-                            AudioRenderer* renderer) = 0;
+                            AudioSource* source) = 0;
   // Gets current energy levels for all incoming streams.
   virtual bool GetActiveStreams(AudioInfo::StreamList* actives) = 0;
   // Get the current energy level of the stream sent to the speaker.
   virtual int GetOutputLevel() = 0;
   // Get the time in milliseconds since last recorded keystroke, or negative.
   virtual int GetTimeSinceLastTyping() = 0;
   // Temporarily exposed field for tuning typing detect options.
   virtual void SetTypingDetectionParameters(int time_window,
     int cost_per_typing, int reporting_threshold, int penalty_decay,
     int type_event_delay) = 0;
@@ skipping 190 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_