diff --git a/audiotools/core/ffmpeg.py b/audiotools/core/ffmpeg.py
index baf27ccc..83f9cd19 100644
--- a/audiotools/core/ffmpeg.py
+++ b/audiotools/core/ffmpeg.py
@@ -176,7 +176,14 @@ def load_from_file_with_ffmpeg(cls, audio_path: str, quiet: bool = True, **kwarg
 
             ff = ffmpy.FFmpeg(
                 inputs={audio_path: None},
-                outputs={wav_file: None},
+                # For inputs that are m4a (and others?), the input audio can
+                # have samples that don't match the sample rate. This aresample
+                # option forces ffmpeg to read timing information in the source
+                # file instead of assuming constant sample rate.
+                #
+                # This fixes an issue where an input m4a file might be a
+                # different length than the output wav file
+                outputs={wav_file: "-af aresample=async=1000"},
                 global_options=global_options,
             )
             ff.run()