[FFmpeg-cvslog] oss_audio: Split muxer and demuxer
Nidhi Makhijani
git at videolan.org
Sat Jul 19 13:45:21 CEST 2014
ffmpeg | branch: master | Nidhi Makhijani <nidhimj22 at gmail.com> | Fri Jul 18 16:31:15 2014 +0530| [d6e1d37100af568211f28ec0bcf7958a3a2a299e] | committer: Diego Biurrun
oss_audio: Split muxer and demuxer
Signed-off-by: Diego Biurrun <diego at biurrun.de>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d6e1d37100af568211f28ec0bcf7958a3a2a299e
---
libavdevice/Makefile | 4 +-
libavdevice/oss_audio.c | 211 +++----------------------------------------
libavdevice/oss_audio.h | 45 +++++++++
libavdevice/oss_audio_dec.c | 146 ++++++++++++++++++++++++++++++
libavdevice/oss_audio_enc.c | 108 ++++++++++++++++++++++
5 files changed, 312 insertions(+), 202 deletions(-)
diff --git a/libavdevice/Makefile b/libavdevice/Makefile
index 2eb2f8e..25e126c 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-$(CONFIG_OSS_INDEV) += oss_audio.o oss_audio_dec.o
+OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o 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
index 95f73fb..ad52d78 100644
--- a/libavdevice/oss_audio.c
+++ b/libavdevice/oss_audio.c
@@ -20,45 +20,31 @@
*/
#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
+#include "libavformat/avformat.h"
-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;
+#include "oss_audio.h"
-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");
@@ -80,7 +66,7 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi
if (!is_output)
fcntl(audio_fd, F_SETFL, O_NONBLOCK);
- s->frame_size = AUDIO_BLOCK_SIZE;
+ s->frame_size = OSS_AUDIO_BLOCK_SIZE;
/* select format : favour native format */
err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
@@ -143,183 +129,8 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi
return AVERROR(EIO);
}
-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,
-};
-
-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..87ac4ad
--- /dev/null
+++ b/libavdevice/oss_audio.h
@@ -0,0 +1,45 @@
+/*
+ * 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 "libavcodec/avcodec.h"
+
+#include "libavformat/avformat.h"
+
+#define OSS_AUDIO_BLOCK_SIZE 4096
+
+typedef struct OSSAudioData {
+ 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[OSS_AUDIO_BLOCK_SIZE];
+ int buffer_ptr;
+} OSSAudioData;
+
+int ff_oss_audio_open(AVFormatContext *s1, int is_output,
+ const char *audio_device);
+
+int ff_oss_audio_close(OSSAudioData *s);
+
+#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..601d91c
--- /dev/null
+++ b/libavdevice/oss_audio_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_audio.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,
+};
diff --git a/libavdevice/oss_audio_enc.c b/libavdevice/oss_audio_enc.c
new file mode 100644
index 0000000..688982a
--- /dev/null
+++ b/libavdevice/oss_audio_enc.c
@@ -0,0 +1,108 @@
+/*
+ * Linux audio 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 <unistd.h>
+#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"
+
+static int audio_write_header(AVFormatContext *s1)
+{
+ OSSAudioData *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_oss_audio_open(s1, 1, s1->filename);
+ if (ret < 0) {
+ return AVERROR(EIO);
+ } else {
+ return 0;
+ }
+}
+
+static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
+{
+ OSSAudioData *s = s1->priv_data;
+ int len, ret;
+ int size= pkt->size;
+ uint8_t *buf= pkt->data;
+
+ while (size > 0) {
+ len = FFMIN(OSS_AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
+ memcpy(s->buffer + s->buffer_ptr, buf, len);
+ s->buffer_ptr += len;
+ if (s->buffer_ptr >= OSS_AUDIO_BLOCK_SIZE) {
+ for(;;) {
+ ret = write(s->fd, s->buffer, OSS_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)
+{
+ OSSAudioData *s = s1->priv_data;
+
+ ff_oss_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(OSSAudioData),
+ /* 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,
+};
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