X-Git-Url: http://git.ffmpeg.org/gitweb/ffmpeg.git/blobdiff_plain/63613fe615160671b394a232c1a3736319a6a8ec..c0cbe36b18ab3eb13a53fe684ec1f63a00df2c86:/libavdevice/oss_audio.c?ds=sidebyside diff --git a/libavdevice/oss_audio.c b/libavdevice/oss_audio.c index 5b0d02a..c19c677 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 #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; @@ -78,13 +80,6 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi fcntl(audio_fd, F_SETFL, O_NONBLOCK); s->frame_size = AUDIO_BLOCK_SIZE; -#if 0 - tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS; - err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp); - if (err < 0) { - perror("SNDCTL_DSP_SETFRAGMENT"); - } -#endif /* select format : favour native format */ err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp); @@ -179,9 +174,7 @@ static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) uint8_t *buf= pkt->data; while (size > 0) { - len = AUDIO_BLOCK_SIZE - s->buffer_ptr; - if (len > size) - len = size; + len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size); memcpy(s->buffer + s->buffer_ptr, buf, len); s->buffer_ptr += len; if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) { @@ -216,24 +209,18 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap) AVStream *st; int ret; - if (ap->sample_rate <= 0 || ap->channels <= 0) - return -1; - 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) { - av_free(st); return AVERROR(EIO); } /* 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; @@ -249,34 +236,15 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) int64_t cur_time; struct audio_buf_info abufi; - if (av_new_packet(pkt, s->frame_size) < 0) - return AVERROR(EIO); - for(;;) { - struct timeval tv; - fd_set fds; - - tv.tv_sec = 0; - tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */ - - FD_ZERO(&fds); - FD_SET(s->fd, &fds); - - /* This will block until data is available or we get a timeout */ - (void) select(s->fd + 1, &fds, 0, 0, &tv); - - ret = read(s->fd, pkt->data, pkt->size); - if (ret > 0) - break; - if (ret == -1 && (errno == EAGAIN || errno == EINTR)) { - av_free_packet(pkt); - pkt->size = 0; - pkt->pts = av_gettime(); - return 0; - } - if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) { - av_free_packet(pkt); - return AVERROR(EIO); - } + if ((ret=av_new_packet(pkt, s->frame_size)) < 0) + return ret; + + 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; } pkt->size = ret; @@ -313,37 +281,44 @@ static int audio_read_close(AVFormatContext *s1) } #if CONFIG_OSS_INDEV -AVInputFormat oss_demuxer = { - "oss", - NULL_IF_CONFIG_SMALL("Open Sound System capture"), - sizeof(AudioData), - NULL, - audio_read_header, - audio_read_packet, - audio_read_close, - .flags = AVFMT_NOFILE, +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 = { + .name = "oss", + .long_name = NULL_IF_CONFIG_SMALL("Open Sound System capture"), + .priv_data_size = sizeof(AudioData), + .read_header = audio_read_header, + .read_packet = audio_read_packet, + .read_close = audio_read_close, + .flags = AVFMT_NOFILE, + .priv_class = &oss_demuxer_class, }; #endif #if CONFIG_OSS_OUTDEV -AVOutputFormat oss_muxer = { - "oss", - NULL_IF_CONFIG_SMALL("Open Sound System playback"), - "", - "", - sizeof(AudioData), +AVOutputFormat ff_oss_muxer = { + .name = "oss", + .long_name = NULL_IF_CONFIG_SMALL("Open Sound System playback"), + .priv_data_size = sizeof(AudioData), /* 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 - CODEC_ID_NONE, - audio_write_header, - audio_write_packet, - audio_write_trailer, - .flags = AVFMT_NOFILE, + .audio_codec = AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE), + .video_codec = CODEC_ID_NONE, + .write_header = audio_write_header, + .write_packet = audio_write_packet, + .write_trailer = audio_write_trailer, + .flags = AVFMT_NOFILE, }; #endif