/*
  * Pulseaudio input
  * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
  *
  * This file is part of Libav.
  *
  * Libav 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.
  *
  * Libav 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 Libav; 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 <lu_zero@gentoo.org>
  */
 
 #include <pulse/simple.h>
 #include <pulse/rtclock.h>
 #include <pulse/error.h>
 
 #include "libavformat/avformat.h"

 #include "libavformat/internal.h"

 #include "libavutil/opt.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 pa_sample_format_t codec_id_to_pulse_format(int codec_id) {

     switch (codec_id) {

     case AV_CODEC_ID_PCM_U8:    return PA_SAMPLE_U8;
     case AV_CODEC_ID_PCM_ALAW:  return PA_SAMPLE_ALAW;
     case AV_CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
     case AV_CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
     case AV_CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
     case AV_CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
     case AV_CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
     case AV_CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
     case AV_CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
     case AV_CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
     case AV_CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;

     default:                 return PA_SAMPLE_INVALID;

     }
 }
 

 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",        "pulse server name",                              OFFSET(server),        AV_OPT_TYPE_STRING, {.str = NULL},     0, 0, D },

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

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

     { "sample_rate",   "sample rate in Hz",                              OFFSET(sample_rate),   AV_OPT_TYPE_INT,    {.i64 = 48000},    1, INT_MAX, D },
     { "channels",      "number of audio channels",                       OFFSET(channels),      AV_OPT_TYPE_INT,    {.i64 = 2},        1, INT_MAX, D },
     { "frame_size",    "number of bytes per frame",                      OFFSET(frame_size),    AV_OPT_TYPE_INT,    {.i64 = 1024},     1, INT_MAX, D },
     { "fragment_size", "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,
 };