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Bug 1651745 - Update logging for MediaEngineWebRTCMicrophoneSource et al. r=padenot 

This patch adds identifiers to existing log messages in the mic source and
AudioInputProcessing, and adds new log messages for complete tracing of frames.

Differential Revision: https://phabricator.services.mozilla.com/D95940
This commit is contained in:
Andreas Pehrson 2020-11-05 16:44:38 +00:00
Родитель 7d63a92c30
Коммит d1cec151fa
1 изменённых файлов: 46 добавлений и 17 удалений
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63
dom/media/webrtc/MediaEngineWebRTCAudio.cpp
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@ -124,8 +124,8 @@ nsresult MediaEngineWebRTCMicrophoneSource::EvaluateSettings(
prefs.mChannels = c.mChannelCount.Get(std::min(prefs.mChannels, maxChannels));
prefs.mChannels = std::max(1, std::min(prefs.mChannels, maxChannels));
LOG("Audio config: aec: %d, agc: %d, noise: %d, channels: %d",
prefs.mAecOn ? prefs.mAec : -1, prefs.mAgcOn ? prefs.mAgc : -1,
LOG("Mic source %p Audio config: aec: %d, agc: %d, noise: %d, channels: %d",
this, prefs.mAecOn ? prefs.mAec : -1, prefs.mAgcOn ? prefs.mAgc : -1,
prefs.mNoiseOn ? prefs.mNoise : -1, prefs.mChannels);
*aOutPrefs = prefs;
@ -484,7 +484,8 @@ nsresult MediaEngineWebRTCMicrophoneSource::Deallocate() {
MOZ_ASSERT(mState != kStarted, "Source not stopped");
mState = kReleased;
LOG("Audio device %s deallocated", NS_ConvertUTF16toUTF8(mDeviceName).get());
LOG("Mic source %p Audio device %s deallocated", this,
NS_ConvertUTF16toUTF8(mDeviceName).get());
return NS_OK;
}
@ -508,7 +509,8 @@ void MediaEngineWebRTCMicrophoneSource::SetTrack(
track->Resume(); // Suspended by MediaManager
}));
LOG("Track %p registered for microphone capture", aTrack.get());
LOG("Mic source %p Track %p registered for microphone capture", this,
aTrack.get());
}
class StartStopMessage : public ControlMessage {
@ -654,6 +656,10 @@ void AudioInputProcessing::SetPassThrough(MediaTrackGraphImpl* aGraph,
MOZ_ASSERT(aGraph->OnGraphThread());
if (!mSkipProcessing && aPassThrough && mPacketizerInput) {
MOZ_ASSERT(mPacketizerInput->PacketsAvailable() == 0);
LOG_FRAME(
"AudioInputProcessing %p Appending %u frames of null data for data "
"discarded in the packetizer",
this, mPacketizerInput->FramesAvailable());
mSegment.AppendNullData(mPacketizerInput->FramesAvailable());
mPacketizerInput->Clear();
}
@ -693,8 +699,10 @@ void AudioInputProcessing::UpdateAECSettings(
if (aLevel != EchoCancellation::SuppressionLevel::kLowSuppression &&
aLevel != EchoCancellation::SuppressionLevel::kModerateSuppression &&
aLevel != EchoCancellation::SuppressionLevel::kHighSuppression) {
LOG_ERROR("Attempt to set invalid AEC suppression level %d",
static_cast<int>(aLevel));
LOG_ERROR(
"AudioInputProcessing %p Attempt to set invalid AEC suppression "
"level %d",
this, static_cast<int>(aLevel));
aLevel = EchoCancellation::SuppressionLevel::kModerateSuppression;
}
@ -711,14 +719,18 @@ void AudioInputProcessing::UpdateAGCSettings(bool aEnable,
if (aMode != GainControl::Mode::kAdaptiveAnalog &&
aMode != GainControl::Mode::kAdaptiveDigital &&
aMode != GainControl::Mode::kFixedDigital) {
LOG_ERROR("Attempt to set invalid AGC mode %d", static_cast<int>(aMode));
LOG_ERROR("AudioInputProcessing %p Attempt to set invalid AGC mode %d",
this, static_cast<int>(aMode));
aMode = GainControl::Mode::kAdaptiveDigital;
}
#if defined(WEBRTC_IOS) || defined(ATA) || defined(WEBRTC_ANDROID)
if (aMode == GainControl::Mode::kAdaptiveAnalog) {
LOG_ERROR("Invalid AGC mode kAgcAdaptiveAnalog on mobile");
LOG_ERROR(
"AudioInputProcessing %p Invalid AGC mode kAgcAdaptiveAnalog on "
"mobile",
this);
MOZ_ASSERT_UNREACHABLE(
"Bad pref set in all.js or in about:config"
" for the auto gain, on mobile.");
@ -739,8 +751,10 @@ void AudioInputProcessing::UpdateNSSettings(
aLevel != NoiseSuppression::Level::kModerate &&
aLevel != NoiseSuppression::Level::kHigh &&
aLevel != NoiseSuppression::Level::kVeryHigh) {
LOG_ERROR("Attempt to set invalid noise suppression level %d",
static_cast<int>(aLevel));
LOG_ERROR(
"AudioInputProcessing %p Attempt to set invalid noise suppression "
"level %d",
this, static_cast<int>(aLevel));
aLevel = NoiseSuppression::Level::kModerate;
}
@ -813,6 +827,9 @@ void AudioInputProcessing::Pull(MediaTrackGraphImpl* aGraph, GraphTime aFrom,
MOZ_ASSERT(mPacketizerInput);
MOZ_ASSERT((buffering - mLiveBufferingAppended) ==
mPacketizerInput->mPacketSize);
LOG_FRAME("AudioInputProcessing %p Inserting %" PRId64
" frames of silence due to buffer increase",
this, buffering - mLiveBufferingAppended);
mSegment.InsertNullDataAtStart(buffering - mLiveBufferingAppended);
mLiveBufferingAppended = buffering;
} else if (MOZ_UNLIKELY(buffering < mLiveBufferingAppended)) {
@ -823,6 +840,9 @@ void AudioInputProcessing::Pull(MediaTrackGraphImpl* aGraph, GraphTime aFrom,
(mLiveBufferingAppended - buffering));
TrackTime frames =
std::min(mSegment.GetDuration(), mLiveBufferingAppended - buffering);
LOG_FRAME("AudioInputProcessing %p Removing %" PRId64
" frames of silence due to buffer decrease",
this, frames);
mLiveBufferingAppended -= frames;
mSegment.RemoveLeading(frames);
}
@ -832,14 +852,18 @@ void AudioInputProcessing::Pull(MediaTrackGraphImpl* aGraph, GraphTime aFrom,
if (!mLiveFramesAppended) {
// First real data being pulled in. Add the appropriate amount of
// buffering before the real data to avoid glitches.
LOG_FRAME("Buffering %" PRId64 " frames of pre-silence for %u channels.",
buffering, mRequestedInputChannelCount);
LOG_FRAME("AudioInputProcessing %p Buffering %" PRId64
" frames of pre-silence for %u channels.",
this, buffering, mRequestedInputChannelCount);
mSegment.InsertNullDataAtStart(buffering - mLiveBufferingAppended);
mLiveFramesAppended = true;
mLiveBufferingAppended = buffering;
}
MOZ_ASSERT(buffering == mLiveBufferingAppended);
TrackTime frames = std::min(mSegment.GetDuration(), delta);
LOG_FRAME("AudioInputProcessing %p Appending %" PRId64
" frames of real data for %u channels.",
this, frames, mRequestedInputChannelCount);
aSegment->AppendSlice(mSegment, 0, frames);
mSegment.RemoveLeading(frames);
delta -= frames;
@ -855,8 +879,9 @@ void AudioInputProcessing::Pull(MediaTrackGraphImpl* aGraph, GraphTime aFrom,
return;
}
LOG_FRAME("Pulling %" PRId64 " frames of silence for %u channels.", delta,
mRequestedInputChannelCount);
LOG_FRAME("AudioInputProcessing %p Pulling %" PRId64
" frames of silence for %u channels.",
this, delta, mRequestedInputChannelCount);
// This assertion fails if we append silence here after having appended live
// frames. Before appending live frames we should add sufficient buffering to
@ -973,6 +998,9 @@ void AudioInputProcessing::PacketizeAndProcess(MediaTrackGraphImpl* aGraph,
aRate / 100, aChannels);
}
LOG_FRAME("AudioInputProcessing %p Appending %zu frames to packetizer", this,
aFrames);
// Packetize our input data into 10ms chunks, deinterleave into planar channel
// buffers, process, and append to the right MediaStreamTrack.
mPacketizerInput->Input(aBuffer, static_cast<uint32_t>(aFrames));
@ -1066,8 +1094,8 @@ void AudioInputProcessing::PacketizeAndProcess(MediaTrackGraphImpl* aGraph,
continue;
}
LOG_FRAME("Appending %" PRIu32 " frames of packetized audio",
mPacketizerInput->mPacketSize);
LOG_FRAME("AudioInputProcessing %p Appending %u frames of packetized audio",
this, mPacketizerInput->mPacketSize);
// We already have planar audio data of the right format. Insert into the
// MTG.
@ -1112,7 +1140,8 @@ void AudioInputProcessing::InsertInGraph(MediaTrackGraphImpl* aGraph,
write_channels.Elements());
}
LOG_FRAME("Appending %zu frames of raw audio", aFrames);
LOG_FRAME("AudioInputProcessing %p Appending %zu frames of raw audio", this,
aFrames);
MOZ_ASSERT(aChannels == channels.Length());
mSegment.AppendFrames(buffer.forget(), channels, aFrames, mPrincipal);