*/
#include "config.h"
-#include <stdlib.h>
-#include <stdio.h>
-#include <stdint.h>
+
#include <string.h>
-#include <errno.h>
+
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif
+
+#if HAVE_UNISTD_H
#include <unistd.h>
+#endif
#include <fcntl.h>
#include <sys/ioctl.h>
-#include "libavutil/internal.h"
#include "libavutil/log.h"
-#include "libavutil/opt.h"
-#include "libavutil/time.h"
+
#include "libavcodec/avcodec.h"
#include "avdevice.h"
-#include "libavformat/internal.h"
-#define AUDIO_BLOCK_SIZE 4096
+#include "oss_audio.h"
-typedef struct {
- AVClass *class;
- int fd;
- int sample_rate;
- int channels;
- int frame_size; /* in bytes ! */
- enum AVCodecID codec_id;
- unsigned int flip_left : 1;
- uint8_t buffer[AUDIO_BLOCK_SIZE];
- int buffer_ptr;
-} AudioData;
-
-static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
+int ff_oss_audio_open(AVFormatContext *s1, int is_output,
+ const char *audio_device)
{
- AudioData *s = s1->priv_data;
+ OSSAudioData *s = s1->priv_data;
int audio_fd;
int tmp, err;
char *flip = getenv("AUDIO_FLIP_LEFT");
}
}
- s->frame_size = AUDIO_BLOCK_SIZE;
+ s->frame_size = OSS_AUDIO_BLOCK_SIZE;
+
+#define CHECK_IOCTL_ERROR(event) \
+ if (err < 0) { \
+ av_log(s1, AV_LOG_ERROR, #event ": %s\n", strerror(errno)); \
+ goto fail; \
+ }
- /* select format : favour native format */
+ /* select format : favour native format
+ * We don't CHECK_IOCTL_ERROR here because even if failed OSS still may be
+ * usable. If OSS is not usable the SNDCTL_DSP_SETFMTS later is going to
+ * fail anyway. `err =` kept to eliminate compiler warning. */
err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
#if HAVE_BIGENDIAN
return AVERROR(EIO);
}
err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
- if (err < 0) {
- av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
- goto fail;
- }
+ CHECK_IOCTL_ERROR(SNDCTL_DSP_SETFMTS)
tmp = (s->channels == 2);
err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
- if (err < 0) {
- av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
- goto fail;
- }
+ CHECK_IOCTL_ERROR(SNDCTL_DSP_STEREO)
tmp = s->sample_rate;
err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
- if (err < 0) {
- av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
- goto fail;
- }
+ CHECK_IOCTL_ERROR(SNDCTL_DSP_SPEED)
s->sample_rate = tmp; /* store real sample rate */
s->fd = audio_fd;
fail:
close(audio_fd);
return AVERROR(EIO);
+#undef CHECK_IOCTL_ERROR
}
-static int audio_close(AudioData *s)
+int ff_oss_audio_close(OSSAudioData *s)
{
close(s->fd);
return 0;
}
-
-/* sound output support */
-static int audio_write_header(AVFormatContext *s1)
-{
- AudioData *s = s1->priv_data;
- AVStream *st;
- int ret;
-
- st = s1->streams[0];
- s->sample_rate = st->codec->sample_rate;
- s->channels = st->codec->channels;
- ret = audio_open(s1, 1, s1->filename);
- if (ret < 0) {
- return AVERROR(EIO);
- } else {
- return 0;
- }
-}
-
-static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
-{
- AudioData *s = s1->priv_data;
- int len, ret;
- int size= pkt->size;
- uint8_t *buf= pkt->data;
-
- while (size > 0) {
- 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) {
- for(;;) {
- ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
- if (ret > 0)
- break;
- if (ret < 0 && (errno != EAGAIN && errno != EINTR))
- return AVERROR(EIO);
- }
- s->buffer_ptr = 0;
- }
- buf += len;
- size -= len;
- }
- return 0;
-}
-
-static int audio_write_trailer(AVFormatContext *s1)
-{
- AudioData *s = s1->priv_data;
-
- audio_close(s);
- return 0;
-}
-
-/* grab support */
-
-static int audio_read_header(AVFormatContext *s1)
-{
- AudioData *s = s1->priv_data;
- AVStream *st;
- int ret;
-
- st = avformat_new_stream(s1, NULL);
- if (!st) {
- return AVERROR(ENOMEM);
- }
-
- ret = audio_open(s1, 0, s1->filename);
- if (ret < 0) {
- return AVERROR(EIO);
- }
-
- /* take real parameters */
- 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;
-
- avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
- return 0;
-}
-
-static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
-{
- AudioData *s = s1->priv_data;
- int ret, bdelay;
- int64_t cur_time;
- struct audio_buf_info abufi;
-
- 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;
-
- /* compute pts of the start of the packet */
- cur_time = av_gettime();
- bdelay = ret;
- if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
- bdelay += abufi.bytes;
- }
- /* subtract time represented by the number of bytes in the audio fifo */
- cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
-
- /* convert to wanted units */
- pkt->pts = cur_time;
-
- if (s->flip_left && s->channels == 2) {
- int i;
- short *p = (short *) pkt->data;
-
- for (i = 0; i < ret; i += 4) {
- *p = ~*p;
- p += 2;
- }
- }
- return 0;
-}
-
-static int audio_read_close(AVFormatContext *s1)
-{
- AudioData *s = s1->priv_data;
-
- audio_close(s);
- return 0;
-}
-
-#if CONFIG_OSS_INDEV
-static const AVOption options[] = {
- { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
- { "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.i64 = 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,
- .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
-};
-
-AVInputFormat ff_oss_demuxer = {
- .name = "oss",
- .long_name = NULL_IF_CONFIG_SMALL("OSS (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
-static const AVClass oss_muxer_class = {
- .class_name = "OSS muxer",
- .item_name = av_default_item_name,
- .version = LIBAVUTIL_VERSION_INT,
- .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
-};
-
-AVOutputFormat ff_oss_muxer = {
- .name = "oss",
- .long_name = NULL_IF_CONFIG_SMALL("OSS (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 */
- .audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
- .video_codec = AV_CODEC_ID_NONE,
- .write_header = audio_write_header,
- .write_packet = audio_write_packet,
- .write_trailer = audio_write_trailer,
- .flags = AVFMT_NOFILE,
- .priv_class = &oss_muxer_class,
-};
-#endif
projects / ffmpeg / blobdiff