Remove Soundclip handling from libjingle.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51009004
Cr-Commit-Position: refs/heads/master@{#9216}
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 242467d..781901e 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -91,7 +91,6 @@
class VoETraceWrapper;
class VoEWrapper;
class VoiceProcessor;
-class WebRtcSoundclipMedia;
class WebRtcVoiceMediaChannel;
// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
@@ -103,9 +102,7 @@
public:
WebRtcVoiceEngine();
// Dependency injection for testing.
- WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
- VoEWrapper* voe_wrapper_sc,
- VoETraceWrapper* tracing);
+ WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
~WebRtcVoiceEngine();
bool Init(rtc::Thread* worker_thread);
void Terminate();
@@ -113,8 +110,6 @@
int GetCapabilities();
VoiceMediaChannel* CreateChannel();
- SoundclipMedia* CreateSoundclip();
-
AudioOptions GetOptions() const { return options_; }
bool SetOptions(const AudioOptions& options);
// Overrides, when set, take precedence over the options on a
@@ -166,21 +161,15 @@
void RegisterChannel(WebRtcVoiceMediaChannel *channel);
void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
- // May only be called by WebRtcSoundclipMedia.
- void RegisterSoundclip(WebRtcSoundclipMedia *channel);
- void UnregisterSoundclip(WebRtcSoundclipMedia *channel);
-
// Called by WebRtcVoiceMediaChannel to set a gain offset from
// the default AGC target level.
bool AdjustAgcLevel(int delta);
VoEWrapper* voe() { return voe_wrapper_.get(); }
- VoEWrapper* voe_sc() { return voe_wrapper_sc_.get(); }
int GetLastEngineError();
- // Set the external ADMs. This can only be called before Init.
- bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
- webrtc::AudioDeviceModule* adm_sc);
+ // Set the external ADM. This can only be called before Init.
+ bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm);
// Starts AEC dump using existing file.
bool StartAecDump(rtc::PlatformFile file);
@@ -190,10 +179,8 @@
// Create a VoiceEngine Channel.
int CreateMediaVoiceChannel();
- int CreateSoundclipVoiceChannel();
private:
- typedef std::vector<WebRtcSoundclipMedia *> SoundclipList;
typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
typedef sigslot::
signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal;
@@ -202,7 +189,6 @@
void ConstructCodecs();
bool GetVoeCodec(int index, webrtc::CodecInst* codec);
bool InitInternal();
- bool EnsureSoundclipEngineInit();
void SetTraceFilter(int filter);
void SetTraceOptions(const std::string& options);
// Applies either options or overrides. Every option that is "set"
@@ -250,13 +236,9 @@
// The primary instance of WebRtc VoiceEngine.
rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
- // A secondary instance, for playing out soundclips (on the 'ring' device).
- rtc::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
- bool voe_wrapper_sc_initialized_;
rtc::scoped_ptr<VoETraceWrapper> tracing_;
// The external audio device manager
webrtc::AudioDeviceModule* adm_;
- webrtc::AudioDeviceModule* adm_sc_;
int log_filter_;
std::string log_options_;
bool is_dumping_aec_;
@@ -264,7 +246,6 @@
std::vector<RtpHeaderExtension> rtp_header_extensions_;
bool desired_local_monitor_enable_;
rtc::scoped_ptr<WebRtcMonitorStream> monitor_;
- SoundclipList soundclips_;
ChannelList channels_;
// channels_ can be read from WebRtc callback thread. We need a lock on that
// callback as well as the RegisterChannel/UnregisterChannel.