/*
  * ALSA input and output
  * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
  * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
  *
  * 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
  */
 
 /**

  * @file

  * ALSA input and output: input
  * @author Luca Abeni ( lucabe72 email it )
  * @author Benoit Fouet ( benoit fouet free fr )
  * @author Nicolas George ( nicolas george normalesup org )
  *

  * This avdevice decoder can capture audio from an ALSA (Advanced

  * Linux Sound Architecture) device.
  *
  * The filename parameter is the name of an ALSA PCM device capable of
  * capture, for example "default" or "plughw:1"; see the ALSA documentation
  * for naming conventions. The empty string is equivalent to "default".
  *
  * The capture period is set to the lower value available for the device,
  * which gives a low latency suitable for real-time capture.
  *
  * The PTS are an Unix time in microsecond.
  *
  * Due to a bug in the ALSA library
  * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
  * decoder does not work with certain ALSA plugins, especially the dsnoop
  * plugin.
  */
 
 #include <alsa/asoundlib.h>

 
 #include "libavutil/internal.h"

 #include "libavutil/mathematics.h"

 #include "libavutil/opt.h"

 #include "libavutil/time.h"

 #include "libavformat/internal.h"

 #include "avdevice.h"

 #include "alsa.h"

 static av_cold int audio_read_header(AVFormatContext *s1)

 {
     AlsaData *s = s1->priv_data;
     AVStream *st;
     int ret;

     enum AVCodecID codec_id;

     st = avformat_new_stream(s1, NULL);

     if (!st) {
         av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
 
         return AVERROR(ENOMEM);
     }

     codec_id    = s1->audio_codec_id;

     ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,

         &codec_id);
     if (ret < 0) {
         return AVERROR(EIO);
     }
 
     /* take real parameters */

     st->codecpar->codec_type  = AVMEDIA_TYPE_AUDIO;
     st->codecpar->codec_id    = codec_id;
     st->codecpar->sample_rate = s->sample_rate;
     st->codecpar->channels    = s->channels;

     st->codecpar->frame_size = s->frame_size;

     avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */

     /* microseconds instead of seconds, MHz instead of Hz */

     s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,

                                       s->period_size, 1.5E-6);

     if (!s->timefilter)
         goto fail;

 
     return 0;
 
 fail:
     snd_pcm_close(s->h);
     return AVERROR(EIO);
 }
 
 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     AlsaData *s  = s1->priv_data;
     int res;

     int64_t dts;
     snd_pcm_sframes_t delay = 0;

     if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {

         return AVERROR(EIO);
     }
 

     while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {

         if (res == -EAGAIN) {

             av_packet_unref(pkt);

 
             return AVERROR(EAGAIN);
         }
         if (ff_alsa_xrun_recover(s1, res) < 0) {
             av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
                    snd_strerror(res));

             av_packet_unref(pkt);

 
             return AVERROR(EIO);
         }

         ff_timefilter_reset(s->timefilter);

     }
 

     dts = av_gettime();
     snd_pcm_delay(s->h, &delay);
     dts -= av_rescale(delay + res, 1000000, s->sample_rate);

     pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
     s->last_period = res;

 
     pkt->size = res * s->frame_size;
 
     return 0;
 }
 

 static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
 {
     return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
 }
 

 static const AVOption options[] = {

     { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
     { "channels",    "", offsetof(AlsaData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },

     { NULL },
 };
 
 static const AVClass alsa_demuxer_class = {

     .class_name     = "ALSA indev",

     .item_name      = av_default_item_name,
     .option         = options,
     .version        = LIBAVUTIL_VERSION_INT,

     .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,

 };
 

 AVInputFormat ff_alsa_demuxer = {

     .name           = "alsa",
     .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio input"),
     .priv_data_size = sizeof(AlsaData),
     .read_header    = audio_read_header,
     .read_packet    = audio_read_packet,
     .read_close     = ff_alsa_close,

     .get_device_list = audio_get_device_list,

     .flags          = AVFMT_NOFILE,
     .priv_class     = &alsa_demuxer_class,

 };