making act segmenter

1 files changed, 327 insertions, 0 deletions

diff --git a/ffmpeg/libavdevice/oss_audio.c b/ffmpeg/libavdevice/oss_audio.c
new file mode 100644
index 0000000..aa40034
--- /dev/null
+++ b/ffmpeg/libavdevice/oss_audio.c
@@ -0,0 +1,327 @@
+/*
+ * Linux audio play and grab interface
+ * Copyright (c) 2000, 2001 Fabrice Bellard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; 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/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
+
+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 = open(audio_device, O_WRONLY);
+    else
+        audio_fd = 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) {
+        if (fcntl(audio_fd, F_SETFL, O_NONBLOCK) < 0) {
+            av_log(s1, AV_LOG_WARNING, "%s: Could not enable non block mode (%s)\n", audio_device, strerror(errno));
+        }
+    }
+
+    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