webrtc/talk/session/media/mediarecorder.h
buildbot@webrtc.org a09a99950e (Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 17:26:08 +00:00

120 lines
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C++

/*
* libjingle
* Copyright 2010 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_MEDIARECORDER_H_
#define TALK_SESSION_MEDIA_MEDIARECORDER_H_
#include <map>
#include <string>
#include "talk/session/media/channel.h"
#include "talk/session/media/mediasink.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/sigslot.h"
namespace rtc {
class Pathname;
class FileStream;
}
namespace cricket {
class BaseChannel;
class VideoChannel;
class VoiceChannel;
class RtpDumpWriter;
// RtpDumpSink implements MediaSinkInterface by dumping the RTP/RTCP packets to
// a file.
class RtpDumpSink : public MediaSinkInterface, public sigslot::has_slots<> {
public:
// Takes ownership of stream.
explicit RtpDumpSink(rtc::StreamInterface* stream);
virtual ~RtpDumpSink();
virtual void SetMaxSize(size_t size);
virtual bool Enable(bool enable);
virtual bool IsEnabled() const { return recording_; }
virtual void OnPacket(const void* data, size_t size, bool rtcp);
virtual void set_packet_filter(int filter);
int packet_filter() const { return packet_filter_; }
void Flush();
private:
size_t max_size_;
bool recording_;
int packet_filter_;
rtc::scoped_ptr<rtc::StreamInterface> stream_;
rtc::scoped_ptr<RtpDumpWriter> writer_;
rtc::CriticalSection critical_section_;
DISALLOW_COPY_AND_ASSIGN(RtpDumpSink);
};
class MediaRecorder {
public:
MediaRecorder();
virtual ~MediaRecorder();
bool AddChannel(VoiceChannel* channel,
rtc::StreamInterface* send_stream,
rtc::StreamInterface* recv_stream,
int filter);
bool AddChannel(VideoChannel* channel,
rtc::StreamInterface* send_stream,
rtc::StreamInterface* recv_stream,
int filter);
void RemoveChannel(BaseChannel* channel, SinkType type);
bool EnableChannel(BaseChannel* channel, bool enable_send, bool enable_recv,
SinkType type);
void FlushSinks();
private:
struct SinkPair {
bool video_channel;
int filter;
rtc::scoped_ptr<RtpDumpSink> send_sink;
rtc::scoped_ptr<RtpDumpSink> recv_sink;
};
bool InternalAddChannel(BaseChannel* channel,
bool video_channel,
rtc::StreamInterface* send_stream,
rtc::StreamInterface* recv_stream,
int filter);
std::map<BaseChannel*, SinkPair*> sinks_;
rtc::CriticalSection critical_section_;
DISALLOW_COPY_AND_ASSIGN(MediaRecorder);
};
} // namespace cricket
#endif // TALK_SESSION_MEDIA_MEDIARECORDER_H_