/*
  * 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 <sys/time.h>

 #include <sys/select.h>

 #include "libavutil/log.h"

 #include "libavutil/opt.h"

 #include "libavcodec/avcodec.h"

 #include "avdevice.h"

 #define AUDIO_BLOCK_SIZE 4096
 

 typedef struct {

     AVClass *class;

     int fd;

     int sample_rate;

     int channels;

     int frame_size; /* in bytes ! */

     enum CodecID 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)
         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);

 #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 = CODEC_ID_PCM_S16LE;
         break;
     case AFMT_S16_BE:
         s->codec_id = 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, AVFormatParameters *ap)
 {

     AudioData *s = s1->priv_data;

     AVStream *st;
     int ret;
 

 #if FF_API_FORMAT_PARAMETERS

     if (ap->sample_rate > 0)
         s->sample_rate = ap->sample_rate;
     if (ap->channels > 0)
         s->channels = ap->channels;

 #endif

     st = av_new_stream(s1, 0);

     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;

     av_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), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
     { "channels",    "", offsetof(AudioData, channels),    FF_OPT_TYPE_INT, {.dbl = 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 = {

     "oss",

     NULL_IF_CONFIG_SMALL("Open Sound System capture"),

     sizeof(AudioData),
     NULL,
     audio_read_header,
     audio_read_packet,
     audio_read_close,

     .flags = AVFMT_NOFILE,

     .priv_class = &oss_demuxer_class,

 };

 #endif

 #if CONFIG_OSS_OUTDEV

 AVOutputFormat ff_oss_muxer = {

     "oss",

     NULL_IF_CONFIG_SMALL("Open Sound System playback"),

     "",
     "",

     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 */

     AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),

     CODEC_ID_NONE,
     audio_write_header,
     audio_write_packet,
     audio_write_trailer,

     .flags = AVFMT_NOFILE,

 };

 #endif