diff audio.c @ 0:05318cf2e886 libavformat

renamed libav to libavformat

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/audio.c	Mon Nov 25 19:07:40 2002 +0000
@@ -0,0 +1,330 @@
+/*
+ * Linux audio play and grab interface
+ * Copyright (c) 2000, 2001 Fabrice Bellard.
+ *
+ * This library 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 of the License, or (at your option) any later version.
+ *
+ * This library 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 this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+#include "avformat.h"
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <sys/soundcard.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/ioctl.h>
+#include <sys/mman.h>
+#include <sys/time.h>
+
+const char *audio_device = "/dev/dsp";
+
+#define AUDIO_BLOCK_SIZE 4096
+
+typedef struct {
+    int fd;
+    int sample_rate;
+    int channels;
+    int frame_size; /* in bytes ! */
+    int codec_id;
+    int flip_left : 1;
+    UINT8 buffer[AUDIO_BLOCK_SIZE];
+    int buffer_ptr;
+} AudioData;
+
+static int audio_open(AudioData *s, int is_output)
+{
+    int audio_fd;
+    int tmp, err;
+    char *flip = getenv("AUDIO_FLIP_LEFT");
+
+    /* open linux audio device */
+    if (is_output)
+        audio_fd = open(audio_device, O_WRONLY);
+    else
+        audio_fd = open(audio_device, O_RDONLY);
+    if (audio_fd < 0) {
+        perror(audio_device);
+        return -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;
+#if 0
+    tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
+    err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
+    if (err < 0) {
+        perror("SNDCTL_DSP_SETFRAGMENT");
+    }
+#endif
+
+    /* select format : favour native format */
+    err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
+    
+#ifdef WORDS_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 = CODEC_ID_PCM_S16LE;
+        break;
+    case AFMT_S16_BE:
+        s->codec_id = CODEC_ID_PCM_S16BE;
+        break;
+    default:
+        fprintf(stderr, "Soundcard does not support 16 bit sample format\n");
+        close(audio_fd);
+        return -EIO;
+    }
+    err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
+    if (err < 0) {
+        perror("SNDCTL_DSP_SETFMT");
+        goto fail;
+    }
+    
+    tmp = (s->channels == 2);
+    err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
+    if (err < 0) {
+        perror("SNDCTL_DSP_STEREO");
+        goto fail;
+    }
+    if (tmp)
+        s->channels = 2;
+    
+    tmp = s->sample_rate;
+    err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
+    if (err < 0) {
+        perror("SNDCTL_DSP_SPEED");
+        goto fail;
+    }
+    s->sample_rate = tmp; /* store real sample rate */
+    s->fd = audio_fd;
+
+    return 0;
+ fail:
+    close(audio_fd);
+    return -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(s, 1);
+    if (ret < 0) {
+        return -EIO;
+    } else {
+        return 0;
+    }
+}
+
+static int audio_write_packet(AVFormatContext *s1, int stream_index,
+                              UINT8 *buf, int size, int force_pts)
+{
+    AudioData *s = s1->priv_data;
+    int len, ret;
+
+    while (size > 0) {
+        len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
+        if (len > size)
+            len = 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 -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, AVFormatParameters *ap)
+{
+    AudioData *s = s1->priv_data;
+    AVStream *st;
+    int ret;
+
+    if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
+        return -1;
+
+    st = av_new_stream(s1, 0);
+    if (!st) {
+        return -ENOMEM;
+    }
+    s->sample_rate = ap->sample_rate;
+    s->channels = ap->channels;
+
+    ret = audio_open(s, 0);
+    if (ret < 0) {
+        av_free(st);
+        return -EIO;
+    }
+
+    /* take real parameters */
+    st->codec.codec_type = CODEC_TYPE_AUDIO;
+    st->codec.codec_id = s->codec_id;
+    st->codec.sample_rate = s->sample_rate;
+    st->codec.channels = s->channels;
+
+    av_set_pts_info(s1, 48, 1, 1000000);  /* 48 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 (av_new_packet(pkt, s->frame_size) < 0)
+        return -EIO;
+    for(;;) {
+        ret = read(s->fd, pkt->data, pkt->size);
+        if (ret > 0)
+            break;
+        if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
+            av_free_packet(pkt);
+            pkt->size = 0;
+            return 0;
+        }
+        if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
+            av_free_packet(pkt);
+            return -EIO;
+        }
+    }
+    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;
+    }
+    /* substract 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 & ((1LL << 48) - 1);
+
+    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;
+}
+
+static AVInputFormat audio_in_format = {
+    "audio_device",
+    "audio grab and output",
+    sizeof(AudioData),
+    NULL,
+    audio_read_header,
+    audio_read_packet,
+    audio_read_close,
+    .flags = AVFMT_NOFILE,
+};
+
+static AVOutputFormat audio_out_format = {
+    "audio_device",
+    "audio grab and output",
+    "",
+    "",
+    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 */
+#ifdef WORDS_BIGENDIAN
+    CODEC_ID_PCM_S16BE,
+#else
+    CODEC_ID_PCM_S16LE,
+#endif
+    CODEC_ID_NONE,
+    audio_write_header,
+    audio_write_packet,
+    audio_write_trailer,
+    .flags = AVFMT_NOFILE,
+};
+
+int audio_init(void)
+{
+    av_register_input_format(&audio_in_format);
+    av_register_output_format(&audio_out_format);
+    return 0;
+}