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(Auto)update libjingle 69188577-> 69260070

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6437 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-06-16 07:11:01 +00:00
parent ab23d493e0
commit 117afeec91
3 changed files with 11 additions and 7 deletions
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1
talk/session/media/call.h
View File

@ -49,6 +49,7 @@
namespace cricket {
struct AudioInfo;
class Call;
class MediaSessionClient;
class BaseChannel;
class VoiceChannel;

4
talk/session/media/currentspeakermonitor.cc
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@ -28,7 +28,9 @@
#include "talk/session/media/currentspeakermonitor.h"
#include "talk/base/logging.h"
#include "talk/session/media/call.h"
#include "talk/media/base/streamparams.h"
#include "talk/session/media/audiomonitor.h"
#include "talk/session/media/mediamessages.h"
namespace cricket {

13
talk/session/media/currentspeakermonitor.h
View File

@ -39,7 +39,6 @@
namespace cricket {
class BaseSession;
class Call;
class Session;
struct AudioInfo;
struct MediaStreams;
@ -66,7 +65,8 @@ class AudioSourceContext {
// It's recommended that the audio monitor be started with a 100 ms period.
class CurrentSpeakerMonitor : public sigslot::has_slots<> {
public:
CurrentSpeakerMonitor(AudioSourceContext* call, BaseSession* session);
CurrentSpeakerMonitor(AudioSourceContext* audio_source_context,
BaseSession* session);
~CurrentSpeakerMonitor();
BaseSession* session() const { return session_; }
@ -80,13 +80,14 @@ class CurrentSpeakerMonitor : public sigslot::has_slots<> {
void set_min_time_between_switches(uint32 min_time_between_switches);
// This is fired when the current speaker changes, and provides his audio
// SSRC. This only fires after the audio monitor on the underlying Call has
// been started.
// SSRC. This only fires after the audio monitor on the underlying
// AudioSourceContext has been started.
sigslot::signal2<CurrentSpeakerMonitor*, uint32> SignalUpdate;
private:
void OnAudioMonitor(AudioSourceContext* call, const AudioInfo& info);
void OnMediaStreamsUpdate(AudioSourceContext* call,
void OnAudioMonitor(AudioSourceContext* audio_source_context,
const AudioInfo& info);
void OnMediaStreamsUpdate(AudioSourceContext* audio_source_context,
BaseSession* session,
const MediaStreams& added,
const MediaStreams& removed);