webrtc/talk/session/media/audiomonitor.h
henrike@webrtc.org 269fb4bc90 move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00

76 lines
2.5 KiB
C++

/*
* libjingle
* Copyright 2004 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_SESSION_MEDIA_AUDIOMONITOR_H_
#define TALK_SESSION_MEDIA_AUDIOMONITOR_H_
#include <vector>
#include "webrtc/p2p/base/port.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h"
namespace cricket {
class VoiceChannel;
struct AudioInfo {
int input_level;
int output_level;
typedef std::vector<std::pair<uint32, int> > StreamList;
StreamList active_streams; // ssrcs contributing to output_level
};
class AudioMonitor : public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
AudioMonitor(VoiceChannel* voice_channel, rtc::Thread *monitor_thread);
~AudioMonitor();
void Start(int cms);
void Stop();
VoiceChannel* voice_channel();
rtc::Thread *monitor_thread();
sigslot::signal2<AudioMonitor*, const AudioInfo&> SignalUpdate;
protected:
void OnMessage(rtc::Message *message);
void PollVoiceChannel();
AudioInfo audio_info_;
VoiceChannel* voice_channel_;
rtc::Thread* monitoring_thread_;
rtc::CriticalSection crit_;
uint32 rate_;
bool monitoring_;
};
}
#endif // TALK_SESSION_MEDIA_AUDIOMONITOR_H_