cosmetics: Remove extra newlines at EOF

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

diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c
index c467fc097f478bbf03396c6e5c52b2292b7b2f78..b6a7ac20feeebb66f14e3b7865b097f7061290d8 100644 (file)
--- a/libavdevice/alsa-audio-dec.c
+++ b/libavdevice/alsa-audio-dec.c
@@ -47,41 +47,28 @@
 
 #include <alsa/asoundlib.h>
 #include "libavformat/avformat.h"
+#include "libavformat/internal.h"
+#include "libavutil/opt.h"
 
 #include "alsa-audio.h"
 
-static av_cold int audio_read_header(AVFormatContext *s1,
-                                     AVFormatParameters *ap)
+static av_cold int audio_read_header(AVFormatContext *s1)
 {
     AlsaData *s = s1->priv_data;
     AVStream *st;
     int ret;
-    unsigned int sample_rate;
     enum CodecID codec_id;
     snd_pcm_sw_params_t *sw_params;
 
-    if (ap->sample_rate <= 0) {
-        av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate);
-
-        return AVERROR(EIO);
-    }
-
-    if (ap->channels <= 0) {
-        av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels);
-
-        return AVERROR(EIO);
-    }
-
-    st = av_new_stream(s1, 0);
+    st = avformat_new_stream(s1, NULL);
     if (!st) {
         av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
 
         return AVERROR(ENOMEM);
     }
-    sample_rate = ap->sample_rate;
     codec_id    = s1->audio_codec_id;
 
-    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels,
+    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
         &codec_id);
     if (ret < 0) {
         return AVERROR(EIO);
@@ -113,9 +100,9 @@ static av_cold int audio_read_header(AVFormatContext *s1,
     /* take real parameters */
     st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
     st->codec->codec_id    = codec_id;
-    st->codec->sample_rate = sample_rate;
-    st->codec->channels    = ap->channels;
-    av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
+    st->codec->sample_rate = s->sample_rate;
+    st->codec->channels    = s->channels;
+    avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
 
     return 0;
 
@@ -163,13 +150,26 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
     return 0;
 }
 
+static const AVOption options[] = {
+    { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+    { "channels",    "", offsetof(AlsaData, channels),    AV_OPT_TYPE_INT, {.dbl = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+    { NULL },
+};
+
+static const AVClass alsa_demuxer_class = {
+    .class_name     = "ALSA demuxer",
+    .item_name      = av_default_item_name,
+    .option         = options,
+    .version        = LIBAVUTIL_VERSION_INT,
+};
+
 AVInputFormat ff_alsa_demuxer = {
-    "alsa",
-    NULL_IF_CONFIG_SMALL("ALSA audio input"),
-    sizeof(AlsaData),
-    NULL,
-    audio_read_header,
-    audio_read_packet,
-    ff_alsa_close,
-    .flags = AVFMT_NOFILE,
+    .name           = "alsa",
+    .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio input"),
+    .priv_data_size = sizeof(AlsaData),
+    .read_header    = audio_read_header,
+    .read_packet    = audio_read_packet,
+    .read_close     = ff_alsa_close,
+    .flags          = AVFMT_NOFILE,
+    .priv_class     = &alsa_demuxer_class,
 };