lavdev/lavfi: add audio support

[ffmpeg] / libavdevice / oss_audio.c

diff --git a/libavdevice/oss_audio.c b/libavdevice/oss_audio.c
index fcbe26ba93c19de7be88ba4de4a3de93772f3ac8..73cb575bb9da89227fe28ce6d93b2ac87aaca571 100644 (file)
--- a/libavdevice/oss_audio.c
+++ b/libavdevice/oss_audio.c
@@ -2,20 +2,20 @@
  * Linux audio play and grab interface
  * Copyright (c) 2000, 2001 Fabrice Bellard
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -39,7 +39,7 @@
 #include "libavutil/log.h"
 #include "libavutil/opt.h"
 #include "libavcodec/avcodec.h"
-#include "libavformat/avformat.h"
+#include "avdevice.h"
 
 #define AUDIO_BLOCK_SIZE 4096
 
@@ -80,13 +80,6 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi
         fcntl(audio_fd, F_SETFL, O_NONBLOCK);
 
     s->frame_size = AUDIO_BLOCK_SIZE;
-#if 0
-    tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
-    err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
-    if (err < 0) {
-        perror("SNDCTL_DSP_SETFRAGMENT");
-    }
-#endif
 
     /* select format : favour native format */
     err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
@@ -216,13 +209,6 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
     AVStream *st;
     int ret;
 
-#if FF_API_FORMAT_PARAMETERS
-    if (ap->sample_rate > 0)
-        s->sample_rate = ap->sample_rate;
-    if (ap->channels > 0)
-        s->channels = ap->channels;
-#endif
-
     st = av_new_stream(s1, 0);
     if (!st) {
         return AVERROR(ENOMEM);