Magic-Mirror/gecko-dev
1
0
Fork 0
mirror of https://github.com/mozilla/gecko-dev.git synced 2024-11-25 13:51:41 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity

Bug 1156472 - Part 2 - Rename MediaEngineWebRTCAudioSource to MediaEngineWebRTCMicrophoneSource. r=jesup

Browse Source 
There are now two different possible audio source, so this was getting confusing.
This commit is contained in:
Paul Adenot 2015-07-24 14:28:16 +02:00
parent 6187533e7d
commit 6c2138c080
3 changed files with 39 additions and 34 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

7
dom/media/webrtc/MediaEngineWebRTC.cpp
View File

@ -358,15 +358,14 @@ MediaEngineWebRTC::EnumerateAudioDevices(dom::MediaSourceEnum aMediaSource,
strcpy(uniqueId,deviceName); // safe given assert and initialization/error-check
}
nsRefPtr<MediaEngineWebRTCAudioSource> aSource;
nsRefPtr<MediaEngineWebRTCMicrophoneSource> aSource;
NS_ConvertUTF8toUTF16 uuid(uniqueId);
if (mAudioSources.Get(uuid, getter_AddRefs(aSource))) {
// We've already seen this device, just append.
aASources->AppendElement(aSource.get());
} else {
aSource = new MediaEngineWebRTCAudioSource(
mThread, mVoiceEngine, i, deviceName, uniqueId
);
aSource = new MediaEngineWebRTCMicrophoneSource(mThread, mVoiceEngine, i,
deviceName, uniqueId);
mAudioSources.Put(uuid, aSource); // Hashtable takes ownership.
aASources->AppendElement(aSource);
}

14
dom/media/webrtc/MediaEngineWebRTC.h
View File

@ -133,13 +133,16 @@ private:
void GetCapability(size_t aIndex, webrtc::CaptureCapability& aOut) override;
};
class MediaEngineWebRTCAudioSource : public MediaEngineAudioSource,
class MediaEngineWebRTCMicrophoneSource : public MediaEngineAudioSource,
public webrtc::VoEMediaProcess,
private MediaConstraintsHelper
{
public:
MediaEngineWebRTCAudioSource(nsIThread* aThread, webrtc::VoiceEngine* aVoiceEnginePtr,
int aIndex, const char* name, const char* uuid)
MediaEngineWebRTCMicrophoneSource(nsIThread* aThread,
webrtc::VoiceEngine* aVoiceEnginePtr,
int aIndex,
const char* name,
const char* uuid)
: MediaEngineAudioSource(kReleased)
, mVoiceEngine(aVoiceEnginePtr)
, mMonitor("WebRTCMic.Monitor")
@ -207,7 +210,7 @@ public:
virtual void Shutdown() override;
protected:
~MediaEngineWebRTCAudioSource() { Shutdown(); }
~MediaEngineWebRTCMicrophoneSource() { Shutdown(); }
private:
void Init();
@ -294,7 +297,8 @@ private:
// Store devices we've already seen in a hashtable for quick return.
// Maps UUID to MediaEngineSource (one set for audio, one for video).
nsRefPtrHashtable<nsStringHashKey, MediaEngineVideoSource> mVideoSources;
nsRefPtrHashtable<nsStringHashKey, MediaEngineWebRTCAudioSource> mAudioSources;
nsRefPtrHashtable<nsStringHashKey, MediaEngineWebRTCMicrophoneSource>
mAudioSources;
};
}

32
dom/media/webrtc/MediaEngineWebRTCAudio.cpp
View File

@ -41,9 +41,9 @@ extern PRLogModuleInfo* GetMediaManagerLog();
#define LOG_FRAMES(msg) MOZ_LOG(GetMediaManagerLog(), mozilla::LogLevel::Verbose, msg)
/**
* Webrtc audio source.
* Webrtc microphone source source.
*/
NS_IMPL_ISUPPORTS0(MediaEngineWebRTCAudioSource)
NS_IMPL_ISUPPORTS0(MediaEngineWebRTCMicrophoneSource)
// XXX temp until MSG supports registration
StaticRefPtr<AudioOutputObserver> gFarendObserver;
@ -177,7 +177,7 @@ AudioOutputObserver::InsertFarEnd(const AudioDataValue *aBuffer, uint32_t aFrame
}
void
MediaEngineWebRTCAudioSource::GetName(nsAString& aName)
MediaEngineWebRTCMicrophoneSource::GetName(nsAString& aName)
{
if (mInitDone) {
aName.Assign(mDeviceName);
@ -187,7 +187,7 @@ MediaEngineWebRTCAudioSource::GetName(nsAString& aName)
}
void
MediaEngineWebRTCAudioSource::GetUUID(nsACString& aUUID)
MediaEngineWebRTCMicrophoneSource::GetUUID(nsACString& aUUID)
{
if (mInitDone) {
aUUID.Assign(mDeviceUUID);
@ -197,7 +197,7 @@ MediaEngineWebRTCAudioSource::GetUUID(nsACString& aUUID)
}
nsresult
MediaEngineWebRTCAudioSource::Config(bool aEchoOn, uint32_t aEcho,
MediaEngineWebRTCMicrophoneSource::Config(bool aEchoOn, uint32_t aEcho,
bool aAgcOn, uint32_t aAGC,
bool aNoiseOn, uint32_t aNoise,
int32_t aPlayoutDelay)
@ -281,7 +281,7 @@ uint32_t MediaEngineWebRTCAudioSource::GetBestFitnessDistance(
}
nsresult
MediaEngineWebRTCAudioSource::Allocate(const dom::MediaTrackConstraints &aConstraints,
MediaEngineWebRTCMicrophoneSource::Allocate(const dom::MediaTrackConstraints &aConstraints,
const MediaEnginePrefs &aPrefs,
const nsString& aDeviceId)
{
@ -309,7 +309,7 @@ MediaEngineWebRTCAudioSource::Allocate(const dom::MediaTrackConstraints &aConstr
}
nsresult
MediaEngineWebRTCAudioSource::Deallocate()
MediaEngineWebRTCMicrophoneSource::Deallocate()
{
bool empty;
{
@ -331,7 +331,8 @@ MediaEngineWebRTCAudioSource::Deallocate()
}
nsresult
MediaEngineWebRTCAudioSource::Start(SourceMediaStream* aStream, TrackID aID)
MediaEngineWebRTCMicrophoneSource::Start(SourceMediaStream *aStream,
TrackID aID)
{
if (!mInitDone || !aStream) {
return NS_ERROR_FAILURE;
@ -384,7 +385,7 @@ MediaEngineWebRTCAudioSource::Start(SourceMediaStream* aStream, TrackID aID)
}
nsresult
MediaEngineWebRTCAudioSource::Stop(SourceMediaStream *aSource, TrackID aID)
MediaEngineWebRTCMicrophoneSource::Stop(SourceMediaStream *aSource, TrackID aID)
{
{
MonitorAutoLock lock(mMonitor);
@ -421,7 +422,7 @@ MediaEngineWebRTCAudioSource::Stop(SourceMediaStream *aSource, TrackID aID)
}
void
MediaEngineWebRTCAudioSource::NotifyPull(MediaStreamGraph* aGraph,
MediaEngineWebRTCMicrophoneSource::NotifyPull(MediaStreamGraph *aGraph,
SourceMediaStream *aSource,
TrackID aID,
StreamTime aDesiredTime)
@ -431,7 +432,7 @@ MediaEngineWebRTCAudioSource::NotifyPull(MediaStreamGraph* aGraph,
}
void
MediaEngineWebRTCAudioSource::Init()
MediaEngineWebRTCMicrophoneSource::Init()
{
mVoEBase = webrtc::VoEBase::GetInterface(mVoiceEngine);
@ -496,7 +497,7 @@ MediaEngineWebRTCAudioSource::Init()
}
void
MediaEngineWebRTCAudioSource::Shutdown()
MediaEngineWebRTCMicrophoneSource::Shutdown()
{
if (!mInitDone) {
// duplicate these here in case we failed during Init()
@ -551,9 +552,10 @@ MediaEngineWebRTCAudioSource::Shutdown()
typedef int16_t sample;
void
MediaEngineWebRTCAudioSource::Process(int channel,
webrtc::ProcessingTypes type, sample* audio10ms,
int length, int samplingFreq, bool isStereo)
MediaEngineWebRTCMicrophoneSource::Process(int channel,
webrtc::ProcessingTypes type,
sample *audio10ms, int length,
int samplingFreq, bool isStereo)
{
// On initial capture, throw away all far-end data except the most recent sample
// since it's already irrelevant and we want to keep avoid confusing the AEC far-end