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Unified Diff: webrtc/modules/audio_processing/audio_buffer.cc

Issue 1862553002: Fix the number of frames used when interleaving in AudioBuffer::InterleaveTo() (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebasing Created 4 years, 8 months ago
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Index: webrtc/modules/audio_processing/audio_buffer.cc
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index f300bb85cd0ba21e4f1913b872dcf0f962caa2f1..f9b9f631d2b813523ed4a9935ecbc3c0af2ab3bd 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -430,10 +430,10 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) {
}
if (frame->num_channels_ == num_channels_) {
- Interleave(data_ptr->ibuf()->channels(), proc_num_frames_, num_channels_,
+ Interleave(data_ptr->ibuf()->channels(), output_num_frames_, num_channels_,
frame->data_);
} else {
- UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], proc_num_frames_,
+ UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], output_num_frames_,
frame->num_channels_, frame->data_);
}
}
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