[libav-devel] [PATCH 4/4] When audio parameters change, do not resample audio if resampling was not done prior to the change and only the sample format changes.

[Justin Ruggles]

This is needed because opening the decoder can change the sample format after
stream initialization.  The resampling criteria should be the same before and
after the change.
---
 ffmpeg.c |    4 ++--
 1 files changed, 2 insertions(+), 2 deletions(-)


diff --git a/ffmpeg.c b/ffmpeg.c
index 9476bdf..a485369 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -768,7 +768,7 @@ need_realloc:
         ffmpeg_exit(1);
     }
 
-    if (enc->channels != dec->channels)
+    if (enc->channels != dec->channels || enc->sample_rate != dec->sample_rate)
         ost->audio_resample = 1;
 
     resample_changed = ost->resample_sample_fmt  != dec->sample_fmt ||
@@ -794,7 +794,7 @@ need_realloc:
             ost->resample_sample_rate == enc->sample_rate) {
             ost->resample = NULL;
             ost->audio_resample = 0;
-        } else {
+        } else if (ost->audio_resample) {
             if (dec->sample_fmt != AV_SAMPLE_FMT_S16)
                 fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n");
             ost->resample = av_audio_resample_init(enc->channels,    dec->channels,

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