Replace all occurrences of AVERROR_IO with AVERROR(EIO).

Originally committed as revision 9760 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 8 insertions, 8 deletions

diff --git a/libavformat/audio.c b/libavformat/audio.c
index b17dcea8e7..ce209d1a9f 100644
--- a/libavformat/audio.c
+++ b/libavformat/audio.c
@@ -59,7 +59,7 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device)
         audio_fd = open(audio_device, O_RDONLY);
     if (audio_fd < 0) {
         perror(audio_device);
-        return AVERROR_IO;
+        return AVERROR(EIO);
     }
 
     if (flip && *flip == '1') {
@@ -110,7 +110,7 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device)
     default:
         av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
         close(audio_fd);
-        return AVERROR_IO;
+        return AVERROR(EIO);
     }
     err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
     if (err < 0) {
@@ -139,7 +139,7 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device)
     return 0;
  fail:
     close(audio_fd);
-    return AVERROR_IO;
+    return AVERROR(EIO);
 }
 
 static int audio_close(AudioData *s)
@@ -160,7 +160,7 @@ static int audio_write_header(AVFormatContext *s1)
     s->channels = st->codec->channels;
     ret = audio_open(s, 1, s1->filename);
     if (ret < 0) {
-        return AVERROR_IO;
+        return AVERROR(EIO);
     } else {
         return 0;
     }
@@ -185,7 +185,7 @@ static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
                 if (ret > 0)
                     break;
                 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
-                    return AVERROR_IO;
+                    return AVERROR(EIO);
             }
             s->buffer_ptr = 0;
         }
@@ -224,7 +224,7 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
     ret = audio_open(s, 0, s1->filename);
     if (ret < 0) {
         av_free(st);
-        return AVERROR_IO;
+        return AVERROR(EIO);
     }
 
     /* take real parameters */
@@ -245,7 +245,7 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
     struct audio_buf_info abufi;
 
     if (av_new_packet(pkt, s->frame_size) < 0)
-        return AVERROR_IO;
+        return AVERROR(EIO);
     for(;;) {
         struct timeval tv;
         fd_set fds;
@@ -270,7 +270,7 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
         }
         if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
             av_free_packet(pkt);
-            return AVERROR_IO;
+            return AVERROR(EIO);
         }
     }
     pkt->size = ret;