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/*

 * Pulseaudio input

 * Copyright (c) 2011 Luca Barbato 

 *

 * 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

 * PulseAudio input using the simple API.

 * @author Luca Barbato 

 */

#include 

#include 

#include 

#include "libavformat/avformat.h"

#include "libavformat/internal.h"

#include "libavutil/opt.h"

#include "pulse_audio_common.h"

#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)

typedef struct PulseData {

    AVClass *class;

    char *server;

    char *name;

    char *stream_name;

    int  sample_rate;

    int  channels;

    int  frame_size;

    int  fragment_size;

    pa_simple *s;

    int64_t pts;

    int64_t frame_duration;

} PulseData;

static av_cold int pulse_read_header(AVFormatContext *s)

{

    PulseData *pd = s->priv_data;

    AVStream *st;

    char *device = NULL;

    int ret;

    enum AVCodecID codec_id =

        s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;

    const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),

                                pd->sample_rate,

                                pd->channels };

    pa_buffer_attr attr = { -1 };

    st = avformat_new_stream(s, NULL);

    if (!st) {

        av_log(s, AV_LOG_ERROR, "Cannot add stream\n");

        return AVERROR(ENOMEM);

    }

    attr.fragsize = pd->fragment_size;

    if (strcmp(s->filename, "default"))

        device = s->filename;

    pd->s = pa_simple_new(pd->server, pd->name,

                          PA_STREAM_RECORD,

                          device, pd->stream_name, &ss,

                          NULL, &attr, &ret);

    if (!pd->s) {

        av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",

               pa_strerror(ret));

        return AVERROR(EIO);

    }

    /* take real parameters */

    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;

    st->codec->codec_id    = codec_id;

    st->codec->sample_rate = pd->sample_rate;

    st->codec->channels    = pd->channels;

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

    pd->pts = AV_NOPTS_VALUE;

    pd->frame_duration = (pd->frame_size * 1000000LL * 8) /

        (pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id));

    return 0;

}

static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)

{

    PulseData *pd  = s->priv_data;

    int res;

    pa_usec_t latency;

    if (av_new_packet(pkt, pd->frame_size) < 0) {

        return AVERROR(ENOMEM);

    }

    if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {

        av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",

               pa_strerror(res));

        av_free_packet(pkt);

        return AVERROR(EIO);

    }

    if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {

        av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",

               pa_strerror(res));

        return AVERROR(EIO);

    }

    if (pd->pts == AV_NOPTS_VALUE) {

        pd->pts = -latency;

    }

    pkt->pts = pd->pts;

    pd->pts += pd->frame_duration;

    return 0;

}

static av_cold int pulse_close(AVFormatContext *s)

{

    PulseData *pd = s->priv_data;

    pa_simple_free(pd->s);

    return 0;

}

#define OFFSET(a) offsetof(PulseData, a)

#define D AV_OPT_FLAG_DECODING_PARAM

static const AVOption options[] = {

    { "server",        "set PulseAudio server",                             OFFSET(server),        AV_OPT_TYPE_STRING, {.str = NULL},     0, 0, D },

    { "name",          "set application name",                              OFFSET(name),          AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT},  0, 0, D },

    { "stream_name",   "set stream description",                            OFFSET(stream_name),   AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },

    { "sample_rate",   "set sample rate in Hz",                             OFFSET(sample_rate),   AV_OPT_TYPE_INT,    {.i64 = 48000},    1, INT_MAX, D },

    { "channels",      "set number of audio channels",                      OFFSET(channels),      AV_OPT_TYPE_INT,    {.i64 = 2},        1, INT_MAX, D },

    { "frame_size",    "set number of bytes per frame",                     OFFSET(frame_size),    AV_OPT_TYPE_INT,    {.i64 = 1024},     1, INT_MAX, D },

    { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT,    {.i64 = -1},      -1, INT_MAX, D },

    { NULL },

};

static const AVClass pulse_demuxer_class = {

    .class_name     = "Pulse demuxer",

    .item_name      = av_default_item_name,

    .option         = options,

    .version        = LIBAVUTIL_VERSION_INT,

};

AVInputFormat ff_pulse_demuxer = {

    .name           = "pulse",

    .long_name      = NULL_IF_CONFIG_SMALL("Pulse audio input"),

    .priv_data_size = sizeof(PulseData),

    .read_header    = pulse_read_header,

    .read_packet    = pulse_read_packet,

    .read_close     = pulse_close,

    .flags          = AVFMT_NOFILE,

    .priv_class     = &pulse_demuxer_class,

};