Minor correction in the docstring of AudioProcessing::set_stream_delay_ms().
BUG=webrtc:7494
NOTRY=True
Review-Url: https://codereview.webrtc.org/2822253002
Cr-Commit-Position: refs/heads/master@{#17762}
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index cc6772b..a1cd398 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -429,7 +429,7 @@
// t_render is the time the first sample of the same frame is rendered by
// the audio hardware.
// - t_capture is the time the first sample of a frame is captured by the
- // audio hardware and t_pull is the time the same frame is passed to
+ // audio hardware and t_process is the time the same frame is passed to
// ProcessStream().
virtual int set_stream_delay_ms(int delay) = 0;
virtual int stream_delay_ms() const = 0;