/*
 * 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 libavdevice/alsa-audio-dec.c
 * 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 allows to 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 "libavformat/avformat.h"
#include <alsa/asoundlib.h>

#include "alsa-audio.h"

av_cold static int audio_read_header(AVFormatContext *s1,
                                     AVFormatParameters *ap)
{
    AlsaData *s = s1->priv_data;
    AVStream *st;
    int ret;
    unsigned int sample_rate;
    int codec_id;
    snd_pcm_sw_params_t *sw_params;

    if (ap->sample_rate <= 0) {
        av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate);

        return AVERROR(EIO);
    }

    if (ap->channels <= 0) {
        av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels);

        return AVERROR(EIO);
    }

    st = av_new_stream(s1, 0);
    if (!st) {
        av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");

        return AVERROR(ENOMEM);
    }
    sample_rate = ap->sample_rate;
    codec_id    = ap->audio_codec_id;

    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels,
        &codec_id);
    if (ret < 0) {
        return AVERROR(EIO);
    }

    if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
        av_log(s1, AV_LOG_WARNING,
               "capture with some ALSA plugins, especially dsnoop, "
               "may hang.\n");

    ret = snd_pcm_sw_params_malloc(&sw_params);
    if (ret < 0) {
        av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
               snd_strerror(ret));
        goto fail;
    }

    snd_pcm_sw_params_current(s->h, sw_params);
    snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);

    ret = snd_pcm_sw_params(s->h, sw_params);
    snd_pcm_sw_params_free(sw_params);
    if (ret < 0) {
        av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
               snd_strerror(ret));
        goto fail;
    }

    /* take real parameters */
    st->codec->codec_type  = CODEC_TYPE_AUDIO;
    st->codec->codec_id    = codec_id;
    st->codec->sample_rate = sample_rate;
    st->codec->channels    = ap->channels;
    av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */

    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;
    AVStream *st = s1->streams[0];
    int res;
    snd_htimestamp_t timestamp;
    snd_pcm_uframes_t ts_delay;

    if (av_new_packet(pkt, s->period_size) < 0) {
        return AVERROR(EIO);
    }

    while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
        if (res == -EAGAIN) {
            av_free_packet(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_free_packet(pkt);

            return AVERROR(EIO);
        }
    }

    snd_pcm_htimestamp(s->h, &ts_delay, &timestamp);
    ts_delay += res;
    pkt->pts = timestamp.tv_sec * 1000000LL
               + (timestamp.tv_nsec * st->codec->sample_rate
                  - ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
               / (st->codec->sample_rate * 1000LL);

    pkt->size = res * s->frame_size;

    return 0;
}

AVInputFormat alsa_demuxer = {
    "alsa",
    NULL_IF_CONFIG_SMALL("ALSA audio input"),
    sizeof(AlsaData),
    NULL,
    audio_read_header,
    audio_read_packet,
    ff_alsa_close,
    .flags = AVFMT_NOFILE,
};