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Bug 1411742 Remove the pref media.getusermedia.playout_delay and the field as they are not used anywhere r=jib

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MozReview-Commit-ID: Gjbeg0zWtJ1

--HG--
extra : rebase_source : 55ad0e8f5e8ec45e7ed365ea516da9ecbc498253
This commit is contained in:
Tom Ritter 2017-11-16 21:57:07 -06:00
parent 3b8bd8d918
commit 5e81122570
8 changed files with 6 additions and 32 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
dom/media/MediaManager.cpp
View File

@ -1807,7 +1807,6 @@ MediaManager::MediaManager()
mPrefs.mAgc = 0;
mPrefs.mNoise = 0;
#endif
mPrefs.mPlayoutDelay = 0;
mPrefs.mFullDuplex = false;
mPrefs.mChannels = 0; // max channels default
nsresult rv;
@ -1820,12 +1819,12 @@ MediaManager::MediaManager()
}
LOG(("%s: default prefs: %dx%d @%dfps, %dHz test tones, aec: %s,"
"agc: %s, noise: %s, aec level: %d, agc level: %d, noise level: %d,"
"playout delay: %d, %sfull_duplex, extended aec %s, delay_agnostic %s "
"%sfull_duplex, extended aec %s, delay_agnostic %s "
"channels %d",
__FUNCTION__, mPrefs.mWidth, mPrefs.mHeight,
mPrefs.mFPS, mPrefs.mFreq, mPrefs.mAecOn ? "on" : "off",
mPrefs.mAgcOn ? "on": "off", mPrefs.mNoiseOn ? "on": "off", mPrefs.mAec,
mPrefs.mAgc, mPrefs.mNoise, mPrefs.mPlayoutDelay, mPrefs.mFullDuplex ? "" : "not ",
mPrefs.mAgc, mPrefs.mNoise, mPrefs.mFullDuplex ? "" : "not ",
mPrefs.mExtendedFilter ? "on" : "off", mPrefs.mDelayAgnostic ? "on" : "off",
mPrefs.mChannels));
}
@ -1897,7 +1896,6 @@ MediaManager::Get() {
prefs->AddObserver("media.getusermedia.agc", sSingleton, false);
prefs->AddObserver("media.getusermedia.noise_enabled", sSingleton, false);
prefs->AddObserver("media.getusermedia.noise", sSingleton, false);
prefs->AddObserver("media.getusermedia.playout_delay", sSingleton, false);
prefs->AddObserver("media.ondevicechange.fakeDeviceChangeEvent.enabled", sSingleton, false);
prefs->AddObserver("media.getusermedia.channels", sSingleton, false);
#endif
@ -3076,7 +3074,6 @@ MediaManager::GetPrefs(nsIPrefBranch *aBranch, const char *aData)
GetPref(aBranch, "media.getusermedia.aec", aData, &mPrefs.mAec);
GetPref(aBranch, "media.getusermedia.agc", aData, &mPrefs.mAgc);
GetPref(aBranch, "media.getusermedia.noise", aData, &mPrefs.mNoise);
GetPref(aBranch, "media.getusermedia.playout_delay", aData, &mPrefs.mPlayoutDelay);
GetPrefBool(aBranch, "media.getusermedia.aec_extended_filter", aData, &mPrefs.mExtendedFilter);
GetPrefBool(aBranch, "media.getusermedia.aec_aec_delay_agnostic", aData, &mPrefs.mDelayAgnostic);
GetPref(aBranch, "media.getusermedia.channels", aData, &mPrefs.mChannels);
@ -3115,7 +3112,6 @@ MediaManager::Shutdown()
prefs->RemoveObserver("media.getusermedia.agc", this);
prefs->RemoveObserver("media.getusermedia.noise_enabled", this);
prefs->RemoveObserver("media.getusermedia.noise", this);
prefs->RemoveObserver("media.getusermedia.playout_delay", this);
prefs->RemoveObserver("media.ondevicechange.fakeDeviceChangeEvent.enabled", this);
prefs->RemoveObserver("media.getusermedia.channels", this);
#endif

2
dom/media/webrtc/MediaEngine.h
View File

@ -94,7 +94,6 @@ public:
, mAec(0)
, mAgc(0)
, mNoise(0)
, mPlayoutDelay(0)
, mFullDuplex(false)
, mExtendedFilter(false)
, mDelayAgnostic(false)
@ -112,7 +111,6 @@ public:
int32_t mAec;
int32_t mAgc;
int32_t mNoise;
int32_t mPlayoutDelay;
bool mFullDuplex;
bool mExtendedFilter;
bool mDelayAgnostic;

1
dom/media/webrtc/MediaEngineWebRTC.h
View File

@ -634,7 +634,6 @@ private:
int32_t mSampleFrequency;
uint64_t mTotalFrames;
uint64_t mLastLogFrames;
int32_t mPlayoutDelay;
NullTransport *mNullTransport;

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

@ -207,7 +207,6 @@ MediaEngineWebRTCMicrophoneSource::MediaEngineWebRTCMicrophoneSource(
, mSampleFrequency(MediaEngine::DEFAULT_SAMPLE_RATE)
, mTotalFrames(0)
, mLastLogFrames(0)
, mPlayoutDelay(0)
, mNullTransport(nullptr)
, mSkipProcessing(false)
, mInputDownmixBuffer(MAX_SAMPLING_FREQ * MAX_CHANNELS / 100)
@ -309,15 +308,12 @@ MediaEngineWebRTCMicrophoneSource::UpdateSingleSource(
// Clamp channelCount to a valid value
prefs.mChannels = std::max(1, std::min(prefs.mChannels, static_cast<int32_t>(maxChannels)));
LOG(("Audio config: aec: %d, agc: %d, noise: %d, delay: %d, channels: %d",
LOG(("Audio config: aec: %d, agc: %d, noise: %d, channels: %d",
prefs.mAecOn ? prefs.mAec : -1,
prefs.mAgcOn ? prefs.mAgc : -1,
prefs.mNoiseOn ? prefs.mNoise : -1,
prefs.mPlayoutDelay,
prefs.mChannels));
mPlayoutDelay = prefs.mPlayoutDelay;
switch (mState) {
case kReleased:
MOZ_ASSERT(aHandle);
@ -969,7 +965,6 @@ MediaEngineWebRTCMicrophoneSource::Process(int channel,
int res = mVoERender->ExternalPlayoutData(buffer->mData,
mAudioOutputObserver->PlayoutFrequency(),
mAudioOutputObserver->PlayoutChannels(),
mPlayoutDelay,
length);
free(buffer);
if (res == -1) {

2
media/webrtc/trunk/webrtc/voice_engine/include/voe_external_media.h
View File

@ -106,7 +106,7 @@ class WEBRTC_DLLEXPORT VoEExternalMedia {
// 48000 kHz sampling rates respectively).
virtual int ExternalPlayoutData(
int16_t speechData10ms[], int samplingFreqHz, int num_channels,
int current_delay_ms, int& lengthSamples) = 0;
int& lengthSamples) = 0;
// This function gets audio for an external playout sink.
// During transmission, this function should be called every ~10 ms

12
media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.cc
View File

@ -265,14 +265,12 @@ int VoEExternalMediaImpl::ExternalPlayoutData(
int16_t speechData10ms[],
int samplingFreqHz,
int num_channels,
int current_delay_ms,
int& lengthSamples)
{
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(shared_->instance_id(), -1),
"ExternalPlayoutData(speechData10ms=0x%x,"
" lengthSamples=%u, samplingFreqHz=%d, current_delay_ms=%d)",
&speechData10ms[0], lengthSamples, samplingFreqHz,
current_delay_ms);
" lengthSamples=%u, samplingFreqHz=%d)",
&speechData10ms[0], lengthSamples, samplingFreqHz);
#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
if (!shared_->statistics().Initialized())
@ -294,12 +292,6 @@ int VoEExternalMediaImpl::ExternalPlayoutData(
"SetExternalRecordingStatus() invalid sample rate");
return -1;
}
if (current_delay_ms < 0)
{
shared_->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
"SetExternalRecordingStatus() invalid delay)");
return -1;
}
// Far-end data is inserted without going through neteq/etc.
// Only supports 10ms chunks; AnalyzeReverseStream() enforces that

1
media/webrtc/trunk/webrtc/voice_engine/voe_external_media_impl.h
View File

@ -41,7 +41,6 @@ class VoEExternalMediaImpl : public VoEExternalMedia {
int16_t speechData10ms[],
int samplingFreqHz,
int num_channels,
int current_delay_ms,
int& lengthSamples) override;
virtual int ExternalPlayoutGetData(int16_t speechData10ms[],

5
modules/libpref/init/all.js
View File

@ -531,27 +531,22 @@ pref("media.getusermedia.agc", 1);
// full_duplex: enable cubeb full-duplex capture/playback
#if defined(XP_MACOSX)
pref("media.peerconnection.capture_delay", 50);
pref("media.getusermedia.playout_delay", 10);
pref("media.navigator.audio.full_duplex", true);
#elif defined(XP_WIN)
pref("media.peerconnection.capture_delay", 50);
pref("media.getusermedia.playout_delay", 40);
pref("media.navigator.audio.full_duplex", true);
#elif defined(ANDROID)
pref("media.peerconnection.capture_delay", 100);
pref("media.getusermedia.playout_delay", 100);
pref("media.navigator.audio.full_duplex", true);
pref("media.navigator.hardware.vp8_encode.acceleration_enabled", true);
pref("media.navigator.hardware.vp8_encode.acceleration_remote_enabled", true);
pref("media.navigator.hardware.vp8_decode.acceleration_enabled", false);
#elif defined(XP_LINUX) || defined(MOZ_SNDIO)
pref("media.peerconnection.capture_delay", 70);
pref("media.getusermedia.playout_delay", 50);
pref("media.navigator.audio.full_duplex", true);
#else
// *BSD, others - merely a guess for now
pref("media.peerconnection.capture_delay", 50);
pref("media.getusermedia.playout_delay", 50);
pref("media.navigator.audio.full_duplex", false);
#endif
// Use MediaDataDecoder API for WebRTC, this includes hardware acceleration for