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ALSA: add channels and sample_rate private options.

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This commit is contained in:
Anton Khirnov 2011-05-23 19:03:10 +02:00
parent 003e63b6df
commit 2ea8faf39f
2 changed files with 26 additions and 15 deletions
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37
libavdevice/alsa-audio-dec.c
View File

@ -47,6 +47,7 @@
#include <alsa/asoundlib.h> #include <alsa/asoundlib.h>
#include "libavformat/avformat.h" #include "libavformat/avformat.h"
#include "libavutil/opt.h"
#include "alsa-audio.h" #include "alsa-audio.h"
@ -56,21 +57,14 @@ static av_cold int audio_read_header(AVFormatContext *s1,
AlsaData *s = s1->priv_data; AlsaData *s = s1->priv_data;
AVStream *st; AVStream *st;
int ret; int ret;
unsigned int sample_rate;
enum CodecID codec_id; enum CodecID codec_id;
snd_pcm_sw_params_t *sw_params; snd_pcm_sw_params_t *sw_params;
if (ap->sample_rate <= 0) { if (ap->sample_rate > 0)
av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate); s->sample_rate = ap->sample_rate;
return AVERROR(EIO); if (ap->channels > 0)
} s->channels = ap->channels;
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 = av_new_stream(s1, 0);
if (!st) { if (!st) {
@ -78,10 +72,9 @@ static av_cold int audio_read_header(AVFormatContext *s1,
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
} }
sample_rate = ap->sample_rate;
codec_id = s1->audio_codec_id; 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); &codec_id);
if (ret < 0) { if (ret < 0) {
return AVERROR(EIO); return AVERROR(EIO);
@ -113,8 +106,8 @@ static av_cold int audio_read_header(AVFormatContext *s1,
/* take real parameters */ /* take real parameters */
st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = codec_id; st->codec->codec_id = codec_id;
st->codec->sample_rate = sample_rate; st->codec->sample_rate = s->sample_rate;
st->codec->channels = ap->channels; st->codec->channels = s->channels;
av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0; return 0;
@ -163,6 +156,19 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
return 0; 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 = { AVInputFormat ff_alsa_demuxer = {
"alsa", "alsa",
NULL_IF_CONFIG_SMALL("ALSA audio input"), NULL_IF_CONFIG_SMALL("ALSA audio input"),
@ -172,4 +178,5 @@ AVInputFormat ff_alsa_demuxer = {
audio_read_packet, audio_read_packet,
ff_alsa_close, ff_alsa_close,
.flags = AVFMT_NOFILE, .flags = AVFMT_NOFILE,
.priv_class = &alsa_demuxer_class,
}; };

4
libavdevice/alsa-audio.h
View File

@ -33,6 +33,7 @@
#include <alsa/asoundlib.h> #include <alsa/asoundlib.h>
#include "config.h" #include "config.h"
#include "libavformat/avformat.h" #include "libavformat/avformat.h"
#include "libavutil/log.h"
/* XXX: we make the assumption that the soundcard accepts this format */ /* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in /* 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) #define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
typedef struct { typedef struct {
AVClass *class;
snd_pcm_t *h; snd_pcm_t *h;
int frame_size; ///< preferred size for reads and writes int frame_size; ///< preferred size for reads and writes
int period_size; ///< bytes per sample * channels int period_size; ///< bytes per sample * channels
int sample_rate; ///< sample rate set by user
int channels; ///< number of channels set by user
} AlsaData; } AlsaData;
/** /**