Modify all codecs to report their supported input and output sample format(s).

Originally committed as revision 14482 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 4 insertions, 0 deletions

diff --git a/libavcodec/libgsm.c b/libavcodec/libgsm.c
index ef2f5e759a..09578b403d 100644
--- a/libavcodec/libgsm.c
+++ b/libavcodec/libgsm.c
@@ -48,6 +48,8 @@ static av_cold int libgsm_init(AVCodecContext *avctx) {
 
         if(!avctx->sample_rate)
             avctx->sample_rate= 8000;
+
+        avctx->sample_fmt = SAMPLE_FMT_S16;
     }else{
         if (avctx->sample_rate != 8000) {
             av_log(avctx, AV_LOG_ERROR, "Sample rate 8000Hz required for GSM, got %dHz\n",
@@ -117,6 +119,7 @@ AVCodec libgsm_encoder = {
     libgsm_init,
     libgsm_encode_frame,
     libgsm_close,
+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM"),
 };
 
@@ -128,6 +131,7 @@ AVCodec libgsm_ms_encoder = {
     libgsm_init,
     libgsm_encode_frame,
     libgsm_close,
+    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("libgsm GSM Microsoft variant"),
 };