oss: Coalesce source files after outdev removal

libavdevice/Makefile

@@ -16,7 +16,7 @@ OBJS-$(CONFIG_BKTR_INDEV)                += bktr.o
 OBJS-$(CONFIG_DV1394_INDEV)              += dv1394.o
 OBJS-$(CONFIG_FBDEV_INDEV)               += fbdev.o
 OBJS-$(CONFIG_JACK_INDEV)                += jack.o timefilter.o
-OBJS-$(CONFIG_OSS_INDEV)                 += oss_dec.o oss.o
+OBJS-$(CONFIG_OSS_INDEV)                 += oss.o
 OBJS-$(CONFIG_PULSE_INDEV)               += pulse.o
 OBJS-$(CONFIG_SNDIO_INDEV)               += sndio.o
 OBJS-$(CONFIG_V4L2_INDEV)                += v4l2.o

libavdevice/oss.c

@@ -28,15 +28,30 @@
 #include <sys/soundcard.h>
 
 #include "libavutil/log.h"
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
 
 #include "libavcodec/avcodec.h"
 
 #include "libavformat/avformat.h"
-
-#include "oss.h"
-
-int ff_oss_audio_open(AVFormatContext *s1, int is_output,
-                      const char *audio_device)
+#include "libavformat/internal.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;
+
+static int oss_audio_open(AVFormatContext *s1, int is_output,
+                          const char *audio_device)
 {
     OSSAudioData *s = s1->priv_data;
     int audio_fd;
@@ -126,8 +141,103 @@ int ff_oss_audio_open(AVFormatContext *s1, int is_output,
 #undef CHECK_IOCTL_ERROR
 }
 
-int ff_oss_audio_close(OSSAudioData *s)
+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 = oss_audio_open(s1, 0, s1->filename);
+    if (ret < 0) {
+        return AVERROR(EIO);
+    }
+
+    /* take real parameters */
+    st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+    st->codecpar->codec_id = s->codec_id;
+    st->codecpar->sample_rate = s->sample_rate;
+    st->codecpar->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_packet_unref(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;
+
     close(s->fd);
     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,
+};

libavdevice/oss.h

-/*
- * 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_H
-#define AVDEVICE_OSS_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_H */

libavdevice/oss_dec.c

-/*
- * 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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
-    st->codecpar->codec_id = s->codec_id;
-    st->codecpar->sample_rate = s->sample_rate;
-    st->codecpar->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_packet_unref(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,
-};