---
libavdevice/Makefile | 4 +-
libavdevice/oss_audio.c | 325 --------------------------------------------
libavdevice/oss_audio.h | 150 ++++++++++++++++++++
libavdevice/oss_audio_dec.c | 144 ++++++++++++++++++++
libavdevice/oss_audio_enc.c | 108 +++++++++++++++
5 files changed, 404 insertions(+), 327 deletions(-)
delete mode 100644 libavdevice/oss_audio.c
create mode 100644 libavdevice/oss_audio.h
create mode 100644 libavdevice/oss_audio_dec.c
create mode 100644 libavdevice/oss_audio_enc.c
diff --git a/libavdevice/Makefile b/libavdevice/Makefile
index 2eb2f8e..7773555 100644
--- a/libavdevice/Makefile
+++ b/libavdevice/Makefile
@@ -15,8 +15,8 @@ OBJS-$(CONFIG_BKTR_INDEV) += bktr.o
OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o
OBJS-$(CONFIG_FBDEV_INDEV) += fbdev.o
OBJS-$(CONFIG_JACK_INDEV) += jack_audio.o timefilter.o
-OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o
-OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o
+OBJS-$(CONFIG_OSS_INDEV) += oss_audio_dec.o
+OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio_enc.o
OBJS-$(CONFIG_PULSE_INDEV) += pulse.o
OBJS-$(CONFIG_SNDIO_INDEV) += sndio_common.o sndio_dec.o
OBJS-$(CONFIG_SNDIO_OUTDEV) += sndio_common.o sndio_enc.o
diff --git a/libavdevice/oss_audio.c b/libavdevice/oss_audio.c
deleted file mode 100644
index f1cc91f..0000000
--- a/libavdevice/oss_audio.c
+++ /dev/null
@@ -1,325 +0,0 @@
-/*
- * Linux audio play and grab 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 <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
-#include <unistd.h>
-#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 "libavformat/avformat.h"
-#include "libavformat/internal.h"
-
-#define AUDIO_BLOCK_SIZE 4096
-
-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)
-{
- AudioData *s = s1->priv_data;
- int audio_fd;
- int tmp, err;
- char *flip = getenv("AUDIO_FLIP_LEFT");
-
- if (is_output)
- audio_fd = avpriv_open(audio_device, O_WRONLY);
- else
- audio_fd = avpriv_open(audio_device, O_RDONLY);
- if (audio_fd < 0) {
- av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
- return AVERROR(EIO);
- }
-
- if (flip && *flip == '1') {
- s->flip_left = 1;
- }
-
- /* non blocking mode */
- if (!is_output)
- fcntl(audio_fd, F_SETFL, O_NONBLOCK);
-
- s->frame_size = AUDIO_BLOCK_SIZE;
-
- /* select format : favour native format */
- err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
-
-#if HAVE_BIGENDIAN
- if (tmp & AFMT_S16_BE) {
- tmp = AFMT_S16_BE;
- } else if (tmp & AFMT_S16_LE) {
- tmp = AFMT_S16_LE;
- } else {
- tmp = 0;
- }
-#else
- if (tmp & AFMT_S16_LE) {
- tmp = AFMT_S16_LE;
- } else if (tmp & AFMT_S16_BE) {
- tmp = AFMT_S16_BE;
- } else {
- tmp = 0;
- }
-#endif
-
- switch(tmp) {
- case AFMT_S16_LE:
- s->codec_id = AV_CODEC_ID_PCM_S16LE;
- break;
- case AFMT_S16_BE:
- s->codec_id = AV_CODEC_ID_PCM_S16BE;
- break;
- default:
- av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample
format\n");
- close(audio_fd);
- 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;
- }
-
- 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;
- }
-
- 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;
- }
- s->sample_rate = tmp; /* store real sample rate */
- s->fd = audio_fd;
-
- return 0;
- fail:
- close(audio_fd);
- return AVERROR(EIO);
-}
-
-static int audio_close(AudioData *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,
-};
-
-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
-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,
-};
-#endif
diff --git a/libavdevice/oss_audio.h b/libavdevice/oss_audio.h
new file mode 100644
index 0000000..abb54d6
--- /dev/null
+++ b/libavdevice/oss_audio.h
@@ -0,0 +1,150 @@
+/*
+ * Linux audio play and grab common header
+ * Copyright (c) 2010 Jacob Meuser
+ *
+ * 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
+ */
+
+#ifndef AVDEVICE_OSS_AUDIO_H
+#define AVDEVICE_OSS_AUDIO_H
+
+#include "config.h"
+
+#if HAVE_SOUNDCARD_H
+#include <soundcard.h>
+#else
+#include <sys/soundcard.h>
+#endif
+
+#include <string.h>
+#include <fcntl.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+
+#include "libavutil/internal.h"
+#include "libavutil/log.h"
+#include "libavcodec/avcodec.h"
+#include "libavformat/avformat.h"
+#include "libavformat/internal.h"
+
+#define AUDIO_BLOCK_SIZE 4096
+
+typedef struct AudioData {
+ 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 ff_audio_open(AVFormatContext *s1, int is_output, const char
*audio_device)
+{
+ AudioData *s = s1->priv_data;
+ int audio_fd;
+ int tmp, err;
+ char *flip = getenv("AUDIO_FLIP_LEFT");
+
+ if (is_output)
+ audio_fd = avpriv_open(audio_device, O_WRONLY);
+ else
+ audio_fd = avpriv_open(audio_device, O_RDONLY);
+ if (audio_fd < 0) {
+ av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
+ return AVERROR(EIO);
+ }
+
+ if (flip && *flip == '1') {
+ s->flip_left = 1;
+ }
+
+ /* non blocking mode */
+ if (!is_output)
+ fcntl(audio_fd, F_SETFL, O_NONBLOCK);
+
+ s->frame_size = AUDIO_BLOCK_SIZE;
+
+ /* select format : favour native format */
+ err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
+
+#if HAVE_BIGENDIAN
+ if (tmp & AFMT_S16_BE) {
+ tmp = AFMT_S16_BE;
+ } else if (tmp & AFMT_S16_LE) {
+ tmp = AFMT_S16_LE;
+ } else {
+ tmp = 0;
+ }
+#else
+ if (tmp & AFMT_S16_LE) {
+ tmp = AFMT_S16_LE;
+ } else if (tmp & AFMT_S16_BE) {
+ tmp = AFMT_S16_BE;
+ } else {
+ tmp = 0;
+ }
+#endif
+
+ switch(tmp) {
+ case AFMT_S16_LE:
+ s->codec_id = AV_CODEC_ID_PCM_S16LE;
+ break;
+ case AFMT_S16_BE:
+ s->codec_id = AV_CODEC_ID_PCM_S16BE;
+ break;
+ default:
+ av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample
format\n");
+ close(audio_fd);
+ 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;
+ }
+
+ 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;
+ }
+
+ 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;
+ }
+ s->sample_rate = tmp; /* store real sample rate */
+ s->fd = audio_fd;
+
+ return 0;
+ fail:
+ close(audio_fd);
+ return AVERROR(EIO);
+}
+
+static int ff_audio_close(AudioData *s)
+{
+ close(s->fd);
+ return 0;
+}
+#endif /* AVDEVICE_OSS_AUDIO_H */
diff --git a/libavdevice/oss_audio_dec.c b/libavdevice/oss_audio_dec.c
new file mode 100644
index 0000000..1d80758
--- /dev/null
+++ b/libavdevice/oss_audio_dec.c
@@ -0,0 +1,144 @@
+/*
+ * Linux audio play and grab 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"
+
+#if HAVE_SOUNDCARD_H
+#include <soundcard.h>
+#else
+#include <sys/soundcard.h>
+#endif
+
+#include <fcntl.h>
+#include <sys/ioctl.h>
+#include <stdint.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_audio.h"
+
+/* 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 = ff_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;
+
+ ff_audio_close(s);
+ return 0;
+}
+
+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,
+};
+
+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,
+};
diff --git a/libavdevice/oss_audio_enc.c b/libavdevice/oss_audio_enc.c
new file mode 100644
index 0000000..6b3001c
--- /dev/null
+++ b/libavdevice/oss_audio_enc.c
@@ -0,0 +1,108 @@
+/*
+ * Linux audio play and grab 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"
+
+#if HAVE_SOUNDCARD_H
+#include <soundcard.h>
+#else
+#include <sys/soundcard.h>
+#endif
+
+#include <fcntl.h>
+#include <sys/ioctl.h>
+
+#include "libavutil/internal.h"
+#include "libavcodec/avcodec.h"
+#include "libavformat/avformat.h"
+#include "libavformat/internal.h"
+
+#include "oss_audio.h"
+
+#define AUDIO_BLOCK_SIZE 4096
+
+/* 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 = ff_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;
+
+ ff_audio_close(s);
+ return 0;
+}
+
+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,
+};
--
1.9.1
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