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libavdevice/oss_audio.c

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00001 /*
00002  * Linux audio play and grab interface
00003  * Copyright (c) 2000, 2001 Fabrice Bellard
00004  *
00005  * This file is part of FFmpeg.
00006  *
00007  * FFmpeg is free software; you can redistribute it and/or
00008  * modify it under the terms of the GNU Lesser General Public
00009  * License as published by the Free Software Foundation; either
00010  * version 2.1 of the License, or (at your option) any later version.
00011  *
00012  * FFmpeg is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00015  * Lesser General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU Lesser General Public
00018  * License along with FFmpeg; if not, write to the Free Software
00019  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00020  */
00021 
00022 #include "config.h"
00023 #include <stdlib.h>
00024 #include <stdio.h>
00025 #include <stdint.h>
00026 #include <string.h>
00027 #include <errno.h>
00028 #if HAVE_SOUNDCARD_H
00029 #include <soundcard.h>
00030 #else
00031 #include <sys/soundcard.h>
00032 #endif
00033 #include <unistd.h>
00034 #include <fcntl.h>
00035 #include <sys/ioctl.h>
00036 #include <sys/time.h>
00037 #include <sys/select.h>
00038 
00039 #include "libavutil/log.h"
00040 #include "libavutil/opt.h"
00041 #include "libavcodec/avcodec.h"
00042 #include "avdevice.h"
00043 
00044 #define AUDIO_BLOCK_SIZE 4096
00045 
00046 typedef struct {
00047     AVClass *class;
00048     int fd;
00049     int sample_rate;
00050     int channels;
00051     int frame_size; /* in bytes ! */
00052     enum CodecID codec_id;
00053     unsigned int flip_left : 1;
00054     uint8_t buffer[AUDIO_BLOCK_SIZE];
00055     int buffer_ptr;
00056 } AudioData;
00057 
00058 static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
00059 {
00060     AudioData *s = s1->priv_data;
00061     int audio_fd;
00062     int tmp, err;
00063     char *flip = getenv("AUDIO_FLIP_LEFT");
00064 
00065     if (is_output)
00066         audio_fd = open(audio_device, O_WRONLY);
00067     else
00068         audio_fd = open(audio_device, O_RDONLY);
00069     if (audio_fd < 0) {
00070         av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
00071         return AVERROR(EIO);
00072     }
00073 
00074     if (flip && *flip == '1') {
00075         s->flip_left = 1;
00076     }
00077 
00078     /* non blocking mode */
00079     if (!is_output)
00080         fcntl(audio_fd, F_SETFL, O_NONBLOCK);
00081 
00082     s->frame_size = AUDIO_BLOCK_SIZE;
00083 #if 0
00084     tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
00085     err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
00086     if (err < 0) {
00087         perror("SNDCTL_DSP_SETFRAGMENT");
00088     }
00089 #endif
00090 
00091     /* select format : favour native format */
00092     err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
00093 
00094 #if HAVE_BIGENDIAN
00095     if (tmp & AFMT_S16_BE) {
00096         tmp = AFMT_S16_BE;
00097     } else if (tmp & AFMT_S16_LE) {
00098         tmp = AFMT_S16_LE;
00099     } else {
00100         tmp = 0;
00101     }
00102 #else
00103     if (tmp & AFMT_S16_LE) {
00104         tmp = AFMT_S16_LE;
00105     } else if (tmp & AFMT_S16_BE) {
00106         tmp = AFMT_S16_BE;
00107     } else {
00108         tmp = 0;
00109     }
00110 #endif
00111 
00112     switch(tmp) {
00113     case AFMT_S16_LE:
00114         s->codec_id = CODEC_ID_PCM_S16LE;
00115         break;
00116     case AFMT_S16_BE:
00117         s->codec_id = CODEC_ID_PCM_S16BE;
00118         break;
00119     default:
00120         av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
00121         close(audio_fd);
00122         return AVERROR(EIO);
00123     }
00124     err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
00125     if (err < 0) {
00126         av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
00127         goto fail;
00128     }
00129 
00130     tmp = (s->channels == 2);
00131     err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
00132     if (err < 0) {
00133         av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
00134         goto fail;
00135     }
00136 
00137     tmp = s->sample_rate;
00138     err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
00139     if (err < 0) {
00140         av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
00141         goto fail;
00142     }
00143     s->sample_rate = tmp; /* store real sample rate */
00144     s->fd = audio_fd;
00145 
00146     return 0;
00147  fail:
00148     close(audio_fd);
00149     return AVERROR(EIO);
00150 }
00151 
00152 static int audio_close(AudioData *s)
00153 {
00154     close(s->fd);
00155     return 0;
00156 }
00157 
00158 /* sound output support */
00159 static int audio_write_header(AVFormatContext *s1)
00160 {
00161     AudioData *s = s1->priv_data;
00162     AVStream *st;
00163     int ret;
00164 
00165     st = s1->streams[0];
00166     s->sample_rate = st->codec->sample_rate;
00167     s->channels = st->codec->channels;
00168     ret = audio_open(s1, 1, s1->filename);
00169     if (ret < 0) {
00170         return AVERROR(EIO);
00171     } else {
00172         return 0;
00173     }
00174 }
00175 
00176 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
00177 {
00178     AudioData *s = s1->priv_data;
00179     int len, ret;
00180     int size= pkt->size;
00181     uint8_t *buf= pkt->data;
00182 
00183     while (size > 0) {
00184         len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
00185         memcpy(s->buffer + s->buffer_ptr, buf, len);
00186         s->buffer_ptr += len;
00187         if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
00188             for(;;) {
00189                 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
00190                 if (ret > 0)
00191                     break;
00192                 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
00193                     return AVERROR(EIO);
00194             }
00195             s->buffer_ptr = 0;
00196         }
00197         buf += len;
00198         size -= len;
00199     }
00200     return 0;
00201 }
00202 
00203 static int audio_write_trailer(AVFormatContext *s1)
00204 {
00205     AudioData *s = s1->priv_data;
00206 
00207     audio_close(s);
00208     return 0;
00209 }
00210 
00211 /* grab support */
00212 
00213 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
00214 {
00215     AudioData *s = s1->priv_data;
00216     AVStream *st;
00217     int ret;
00218 
00219 #if FF_API_FORMAT_PARAMETERS
00220     if (ap->sample_rate > 0)
00221         s->sample_rate = ap->sample_rate;
00222     if (ap->channels > 0)
00223         s->channels = ap->channels;
00224 #endif
00225 
00226     st = av_new_stream(s1, 0);
00227     if (!st) {
00228         return AVERROR(ENOMEM);
00229     }
00230 
00231     ret = audio_open(s1, 0, s1->filename);
00232     if (ret < 0) {
00233         return AVERROR(EIO);
00234     }
00235 
00236     /* take real parameters */
00237     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
00238     st->codec->codec_id = s->codec_id;
00239     st->codec->sample_rate = s->sample_rate;
00240     st->codec->channels = s->channels;
00241 
00242     av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
00243     return 0;
00244 }
00245 
00246 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
00247 {
00248     AudioData *s = s1->priv_data;
00249     int ret, bdelay;
00250     int64_t cur_time;
00251     struct audio_buf_info abufi;
00252 
00253     if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
00254         return ret;
00255 
00256     ret = read(s->fd, pkt->data, pkt->size);
00257     if (ret <= 0){
00258         av_free_packet(pkt);
00259         pkt->size = 0;
00260         if (ret<0)  return AVERROR(errno);
00261         else        return AVERROR_EOF;
00262     }
00263     pkt->size = ret;
00264 
00265     /* compute pts of the start of the packet */
00266     cur_time = av_gettime();
00267     bdelay = ret;
00268     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
00269         bdelay += abufi.bytes;
00270     }
00271     /* subtract time represented by the number of bytes in the audio fifo */
00272     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
00273 
00274     /* convert to wanted units */
00275     pkt->pts = cur_time;
00276 
00277     if (s->flip_left && s->channels == 2) {
00278         int i;
00279         short *p = (short *) pkt->data;
00280 
00281         for (i = 0; i < ret; i += 4) {
00282             *p = ~*p;
00283             p += 2;
00284         }
00285     }
00286     return 0;
00287 }
00288 
00289 static int audio_read_close(AVFormatContext *s1)
00290 {
00291     AudioData *s = s1->priv_data;
00292 
00293     audio_close(s);
00294     return 0;
00295 }
00296 
00297 #if CONFIG_OSS_INDEV
00298 static const AVOption options[] = {
00299     { "sample_rate", "", offsetof(AudioData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
00300     { "channels",    "", offsetof(AudioData, channels),    FF_OPT_TYPE_INT, {.dbl = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
00301     { NULL },
00302 };
00303 
00304 static const AVClass oss_demuxer_class = {
00305     .class_name     = "OSS demuxer",
00306     .item_name      = av_default_item_name,
00307     .option         = options,
00308     .version        = LIBAVUTIL_VERSION_INT,
00309 };
00310 
00311 AVInputFormat ff_oss_demuxer = {
00312     "oss",
00313     NULL_IF_CONFIG_SMALL("Open Sound System capture"),
00314     sizeof(AudioData),
00315     NULL,
00316     audio_read_header,
00317     audio_read_packet,
00318     audio_read_close,
00319     .flags = AVFMT_NOFILE,
00320     .priv_class = &oss_demuxer_class,
00321 };
00322 #endif
00323 
00324 #if CONFIG_OSS_OUTDEV
00325 AVOutputFormat ff_oss_muxer = {
00326     "oss",
00327     NULL_IF_CONFIG_SMALL("Open Sound System playback"),
00328     "",
00329     "",
00330     sizeof(AudioData),
00331     /* XXX: we make the assumption that the soundcard accepts this format */
00332     /* XXX: find better solution with "preinit" method, needed also in
00333        other formats */
00334     AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
00335     CODEC_ID_NONE,
00336     audio_write_header,
00337     audio_write_packet,
00338     audio_write_trailer,
00339     .flags = AVFMT_NOFILE,
00340 };
00341 #endif

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