Index: webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc |
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc |
index 1e679af914b3664746971037a6a2b7cc7a60f0bc..8e7351d033a0bf934bc6d29e95c3cf4c5e5c3a65 100644 |
--- a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc |
+++ b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc |
@@ -41,12 +41,15 @@ const size_t rampSize = sizeof(rampArray)/sizeof(rampArray[0]); |
namespace webrtc { |
uint32_t CalculateEnergy(const AudioFrame& audioFrame) |
{ |
+ if (audioFrame.muted()) return 0; |
+ |
uint32_t energy = 0; |
+ const int16_t* frame_data = audioFrame.data(); |
for(size_t position = 0; position < audioFrame.samples_per_channel_; |
position++) |
{ |
// TODO(andrew): this can easily overflow. |
- energy += audioFrame.data_[position] * audioFrame.data_[position]; |
+ energy += frame_data[position] * frame_data[position]; |
} |
return energy; |
} |
@@ -54,24 +57,29 @@ uint32_t CalculateEnergy(const AudioFrame& audioFrame) |
void RampIn(AudioFrame& audioFrame) |
{ |
assert(rampSize <= audioFrame.samples_per_channel_); |
+ if (audioFrame.muted()) return; |
+ |
+ int16_t* frame_data = audioFrame.mutable_data(); |
for(size_t i = 0; i < rampSize; i++) |
{ |
- audioFrame.data_[i] = static_cast<int16_t>(rampArray[i] * |
- audioFrame.data_[i]); |
+ frame_data[i] = static_cast<int16_t>(rampArray[i] * frame_data[i]); |
} |
} |
void RampOut(AudioFrame& audioFrame) |
{ |
assert(rampSize <= audioFrame.samples_per_channel_); |
+ if (audioFrame.muted()) return; |
+ |
+ int16_t* frame_data = audioFrame.mutable_data(); |
for(size_t i = 0; i < rampSize; i++) |
{ |
const size_t rampPos = rampSize - 1 - i; |
- audioFrame.data_[i] = static_cast<int16_t>(rampArray[rampPos] * |
- audioFrame.data_[i]); |
+ frame_data[i] = static_cast<int16_t>(rampArray[rampPos] * |
+ frame_data[i]); |
} |
- memset(&audioFrame.data_[rampSize], 0, |
+ memset(&frame_data[rampSize], 0, |
(audioFrame.samples_per_channel_ - rampSize) * |
- sizeof(audioFrame.data_[0])); |
+ sizeof(frame_data[0])); |
} |
} // namespace webrtc |