X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavdevice%2Foss_audio.c;h=af46ea890b68d42ce61091b48737a2b43a6e11fa;hb=8fe7b6443fe8721215f1abac3b854cd52092c909;hp=f3edbdbef0ccb01685ed31162bbfa27544ca89d6;hpb=d3298350bba0d5bdb43485356aa752528ec48d11;p=ffmpeg

diff --git a/libavdevice/oss_audio.c b/libavdevice/oss_audio.c
index f3edbdbef0c..af46ea890b6 100644
--- 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 FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -37,12 +37,14 @@
 #include <sys/select.h>
 
 #include "libavutil/log.h"
+#include "libavutil/opt.h"
 #include "libavcodec/avcodec.h"
 #include "libavformat/avformat.h"
 
 #define AUDIO_BLOCK_SIZE 4096
 
 typedef struct {
+    AVClass *class;
     int fd;
     int sample_rate;
     int channels;
@@ -216,15 +218,17 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
     AVStream *st;
     int ret;
 
-    if (ap->sample_rate <= 0 || ap->channels <= 0)
-        return -1;
+#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);
     }
-    s->sample_rate = ap->sample_rate;
-    s->channels = ap->channels;
 
     ret = audio_open(s1, 0, s1->filename);
     if (ret < 0) {
@@ -232,7 +236,7 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
     }
 
     /* take real parameters */
-    st->codec->codec_type = CODEC_TYPE_AUDIO;
+    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
     st->codec->codec_id = s->codec_id;
     st->codec->sample_rate = s->sample_rate;
     st->codec->channels = s->channels;
@@ -251,12 +255,12 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
     if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
         return ret;
 
-        ret = read(s->fd, pkt->data, pkt->size);
+    ret = read(s->fd, pkt->data, pkt->size);
     if (ret <= 0){
         av_free_packet(pkt);
         pkt->size = 0;
         if (ret<0)  return AVERROR(errno);
-        else        return AVERROR(EOF);
+        else        return AVERROR_EOF;
     }
     pkt->size = ret;
 
@@ -293,7 +297,20 @@ static int audio_read_close(AVFormatContext *s1)
 }
 
 #if CONFIG_OSS_INDEV
-AVInputFormat oss_demuxer = {
+static const AVOption options[] = {
+    { "sample_rate", "", offsetof(AudioData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+    { "channels",    "", offsetof(AudioData, channels),    FF_OPT_TYPE_INT, {.dbl = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+    { NULL },
+};
+
+static const AVClass oss_demuxer_class = {
+    .class_name     = "OSS demuxer",
+    .item_name      = av_default_item_name,
+    .option         = options,
+    .version        = LIBAVUTIL_VERSION_INT,
+};
+
+AVInputFormat ff_oss_demuxer = {
     "oss",
     NULL_IF_CONFIG_SMALL("Open Sound System capture"),
     sizeof(AudioData),
@@ -302,11 +319,12 @@ AVInputFormat oss_demuxer = {
     audio_read_packet,
     audio_read_close,
     .flags = AVFMT_NOFILE,
+    .priv_class = &oss_demuxer_class,
 };
 #endif
 
 #if CONFIG_OSS_OUTDEV
-AVOutputFormat oss_muxer = {
+AVOutputFormat ff_oss_muxer = {
     "oss",
     NULL_IF_CONFIG_SMALL("Open Sound System playback"),
     "",
@@ -315,11 +333,7 @@ AVOutputFormat oss_muxer = {
     /* XXX: we make the assumption that the soundcard accepts this format */
     /* XXX: find better solution with "preinit" method, needed also in
        other formats */
-#if HAVE_BIGENDIAN
-    CODEC_ID_PCM_S16BE,
-#else
-    CODEC_ID_PCM_S16LE,
-#endif
+    AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
     CODEC_ID_NONE,
     audio_write_header,
     audio_write_packet,