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-rw-r--r--libavdevice/oss_dec.c146
1 files changed, 146 insertions, 0 deletions
diff --git a/libavdevice/oss_dec.c b/libavdevice/oss_dec.c
new file mode 100644
index 0000000000..3f786cc356
--- /dev/null
+++ b/libavdevice/oss_dec.c
@@ -0,0 +1,146 @@
+/*
+ * Linux audio play interface
+ * Copyright (c) 2000, 2001 Fabrice Bellard
+ *
+ * This file is part of Libav.
+ *
+ * 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.
+ *
+ * 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "config.h"
+
+#include <stdint.h>
+
+#if HAVE_SOUNDCARD_H
+#include <soundcard.h>
+#else
+#include <sys/soundcard.h>
+#endif
+
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/ioctl.h>
+
+#include "libavutil/internal.h"
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+
+#include "libavcodec/avcodec.h"
+
+#include "libavformat/avformat.h"
+#include "libavformat/internal.h"
+
+#include "oss.h"
+
+static int audio_read_header(AVFormatContext *s1)
+{
+ OSSAudioData *s = s1->priv_data;
+ AVStream *st;
+ int ret;
+
+ st = avformat_new_stream(s1, NULL);
+ if (!st) {
+ return AVERROR(ENOMEM);
+ }
+
+ ret = ff_oss_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)
+{
+ OSSAudioData *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)
+{
+ OSSAudioData *s = s1->priv_data;
+
+ ff_oss_audio_close(s);
+ return 0;
+}
+
+static const AVOption options[] = {
+ { "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+ { "channels", "", offsetof(OSSAudioData, 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,
+};
+
+AVInputFormat ff_oss_demuxer = {
+ .name = "oss",
+ .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
+ .priv_data_size = sizeof(OSSAudioData),
+ .read_header = audio_read_header,
+ .read_packet = audio_read_packet,
+ .read_close = audio_read_close,
+ .flags = AVFMT_NOFILE,
+ .priv_class = &oss_demuxer_class,
+};