FFmpeg : Close input file and free any related memory if

[ffmpeg] / libavdevice / alsa-audio-dec.c

diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c
index 48ed75ee1f8440a11fd4a96d99656194f903883e..f1dd29bafc90cd4c1a83907f240b151cc4fa228b 100644 (file)
--- a/libavdevice/alsa-audio-dec.c
+++ b/libavdevice/alsa-audio-dec.c
@@ -21,7 +21,7 @@
  */
 
 /**
- * @file libavdevice/alsa-audio-dec.c
+ * @file
  * ALSA input and output: input
  * @author Luca Abeni ( lucabe72 email it )
  * @author Benoit Fouet ( benoit fouet free fr )
@@ -45,12 +45,12 @@
  * plugin.
  */
 
-#include "libavformat/avformat.h"
 #include <alsa/asoundlib.h>
+#include "libavformat/avformat.h"
 
 #include "alsa-audio.h"
 
-av_cold static int audio_read_header(AVFormatContext *s1,
+static av_cold int audio_read_header(AVFormatContext *s1,
                                      AVFormatParameters *ap)
 {
     AlsaData *s = s1->priv_data;
@@ -79,7 +79,7 @@ av_cold static int audio_read_header(AVFormatContext *s1,
         return AVERROR(ENOMEM);
     }
     sample_rate = ap->sample_rate;
-    codec_id    = ap->audio_codec_id;
+    codec_id    = s1->audio_codec_id;
 
     ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels,
         &codec_id);
@@ -111,7 +111,7 @@ av_cold static int audio_read_header(AVFormatContext *s1,
     }
 
     /* take real parameters */
-    st->codec->codec_type  = CODEC_TYPE_AUDIO;
+    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
     st->codec->codec_id    = codec_id;
     st->codec->sample_rate = sample_rate;
     st->codec->channels    = ap->channels;