From 2ea8faf39ff6f21c2faaf8f9bd060a6636ea65fc Mon Sep 17 00:00:00 2001
From: Anton Khirnov <anton@khirnov.net>
Date: Mon, 23 May 2011 19:03:10 +0200
Subject: [PATCH] ALSA: add channels and sample_rate private options.

---
 libavdevice/alsa-audio-dec.c | 37 +++++++++++++++++++++---------------
 libavdevice/alsa-audio.h     |  4 ++++
 2 files changed, 26 insertions(+), 15 deletions(-)

diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c
index c467fc097f..285d338ff5 100644
--- a/libavdevice/alsa-audio-dec.c
+++ b/libavdevice/alsa-audio-dec.c
@@ -47,6 +47,7 @@
 
 #include <alsa/asoundlib.h>
 #include "libavformat/avformat.h"
+#include "libavutil/opt.h"
 
 #include "alsa-audio.h"
 
@@ -56,21 +57,14 @@ 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);
+    if (ap->sample_rate > 0)
+        s->sample_rate = 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);
-    }
+    if (ap->channels > 0)
+        s->channels = ap->channels;
 
     st = av_new_stream(s1, 0);
     if (!st) {
@@ -78,10 +72,9 @@ static av_cold int audio_read_header(AVFormatContext *s1,
 
         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);
@@ -113,8 +106,8 @@ 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;
+    st->codec->sample_rate = s->sample_rate;
+    st->codec->channels    = s->channels;
     av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
 
     return 0;
@@ -163,6 +156,19 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
     return 0;
 }
 
+static const AVOption options[] = {
+    { "sample_rate", "", offsetof(AlsaData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+    { "channels",    "", offsetof(AlsaData, channels),    FF_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"),
@@ -172,4 +178,5 @@ AVInputFormat ff_alsa_demuxer = {
     audio_read_packet,
     ff_alsa_close,
     .flags = AVFMT_NOFILE,
+    .priv_class = &alsa_demuxer_class,
 };
diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h
index 7a1b01811b..32c07426ef 100644
--- a/libavdevice/alsa-audio.h
+++ b/libavdevice/alsa-audio.h
@@ -33,6 +33,7 @@
 #include <alsa/asoundlib.h>
 #include "config.h"
 #include "libavformat/avformat.h"
+#include "libavutil/log.h"
 
 /* XXX: we make the assumption that the soundcard accepts this format */
 /* XXX: find better solution with "preinit" method, needed also in
@@ -40,9 +41,12 @@
 #define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
 
 typedef struct {
+    AVClass *class;
     snd_pcm_t *h;
     int frame_size;  ///< preferred size for reads and writes
     int period_size; ///< bytes per sample * channels
+    int sample_rate; ///< sample rate set by user
+    int channels;    ///< number of channels set by user
 } AlsaData;
 
 /**