Minor correction of AudioProcessing::set_stream_delay_ms() docstring

webrtc/modules/audio_processing/include/audio_processing.h

Index: webrtc/modules/audio_processing/include/audio_processing.h
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index cc6772bf718fac20351c3b55a1b92c5970e0809a..a1cd398b022f225adc233f7ddaa6abcb66ccd985 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -429,7 +429,7 @@ class AudioProcessing {
   //     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;