diff options
Diffstat (limited to 'libavdevice')
| -rw-r--r-- | libavdevice/alsa-audio-dec.c | 37 | ||||
| -rw-r--r-- | 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; /** |