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Move capture level computation after all processing.

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BUG=issue1065

Review URL: https://webrtc-codereview.appspot.com/930014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3057 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2012-11-07 19:08:03 +00:00
parent 7096fc0126
commit c862f49b2d
1 changed files with 2 additions and 11 deletions
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13
webrtc/voice_engine/transmit_mixer.cc
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@ -351,7 +351,6 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
CheckForSendCodecChanges();
// --- Resample input audio and create/store the initial audio frame
if (GenerateAudioFrame(static_cast<const WebRtc_Word16*>(audioSamples),
nSamples,
nChannels,
@ -372,7 +371,6 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
}
// --- Near-end Voice Quality Enhancement (APM) processing
APMProcessStream(totalDelayMS, clockDrift, currentMicLevel);
if (swap_stereo_channels_ && stereo_codec_)
@ -380,13 +378,11 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
AudioFrameOperations::SwapStereoChannels(&_audioFrame);
// --- Annoying typing detection (utilizes the APM/VAD decision)
#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
TypingDetection();
#endif
// --- Mute during DTMF tone if direct feedback is enabled
if (_remainingMuteMicTimeMs > 0)
{
AudioFrameOperations::Mute(_audioFrame);
@ -398,25 +394,18 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
}
// --- Mute signal
if (_mute)
{
AudioFrameOperations::Mute(_audioFrame);
}
// --- Measure audio level of speech after APM processing
_audioLevel.ComputeLevel(_audioFrame);
// --- Mix with file (does not affect the mixing frequency)
if (_filePlaying)
{
MixOrReplaceAudioWithFile(_mixingFrequency);
}
// --- Record to file
if (_fileRecording)
{
RecordAudioToFile(_mixingFrequency);
@ -433,6 +422,8 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
}
}
// --- Measure audio level of speech after all processing.
_audioLevel.ComputeLevel(_audioFrame);
return 0;
}