Subversion Repositories Kolibri OS

Rev

Blame | Last modification | View Log | RSS feed

  1. /*
  2.  * Pulseaudio input
  3.  * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. /**
  23.  * @file
  24.  * PulseAudio input using the simple API.
  25.  * @author Luca Barbato <lu_zero@gentoo.org>
  26.  */
  27.  
  28. #include <pulse/simple.h>
  29. #include <pulse/rtclock.h>
  30. #include <pulse/error.h>
  31. #include "libavformat/avformat.h"
  32. #include "libavformat/internal.h"
  33. #include "libavutil/opt.h"
  34. #include "pulse_audio_common.h"
  35.  
  36. #define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
  37.  
  38. typedef struct PulseData {
  39.     AVClass *class;
  40.     char *server;
  41.     char *name;
  42.     char *stream_name;
  43.     int  sample_rate;
  44.     int  channels;
  45.     int  frame_size;
  46.     int  fragment_size;
  47.     pa_simple *s;
  48.     int64_t pts;
  49.     int64_t frame_duration;
  50. } PulseData;
  51.  
  52. static av_cold int pulse_read_header(AVFormatContext *s)
  53. {
  54.     PulseData *pd = s->priv_data;
  55.     AVStream *st;
  56.     char *device = NULL;
  57.     int ret;
  58.     enum AVCodecID codec_id =
  59.         s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
  60.     const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),
  61.                                 pd->sample_rate,
  62.                                 pd->channels };
  63.  
  64.     pa_buffer_attr attr = { -1 };
  65.  
  66.     st = avformat_new_stream(s, NULL);
  67.  
  68.     if (!st) {
  69.         av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
  70.         return AVERROR(ENOMEM);
  71.     }
  72.  
  73.     attr.fragsize = pd->fragment_size;
  74.  
  75.     if (strcmp(s->filename, "default"))
  76.         device = s->filename;
  77.  
  78.     pd->s = pa_simple_new(pd->server, pd->name,
  79.                           PA_STREAM_RECORD,
  80.                           device, pd->stream_name, &ss,
  81.                           NULL, &attr, &ret);
  82.  
  83.     if (!pd->s) {
  84.         av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
  85.                pa_strerror(ret));
  86.         return AVERROR(EIO);
  87.     }
  88.     /* take real parameters */
  89.     st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
  90.     st->codec->codec_id    = codec_id;
  91.     st->codec->sample_rate = pd->sample_rate;
  92.     st->codec->channels    = pd->channels;
  93.     avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
  94.  
  95.     pd->pts = AV_NOPTS_VALUE;
  96.     pd->frame_duration = (pd->frame_size * 1000000LL * 8) /
  97.         (pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id));
  98.  
  99.     return 0;
  100. }
  101.  
  102. static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
  103. {
  104.     PulseData *pd  = s->priv_data;
  105.     int res;
  106.     pa_usec_t latency;
  107.  
  108.     if (av_new_packet(pkt, pd->frame_size) < 0) {
  109.         return AVERROR(ENOMEM);
  110.     }
  111.  
  112.     if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
  113.         av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
  114.                pa_strerror(res));
  115.         av_free_packet(pkt);
  116.         return AVERROR(EIO);
  117.     }
  118.  
  119.     if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
  120.         av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
  121.                pa_strerror(res));
  122.         return AVERROR(EIO);
  123.     }
  124.  
  125.     if (pd->pts == AV_NOPTS_VALUE) {
  126.         pd->pts = -latency;
  127.     }
  128.  
  129.     pkt->pts = pd->pts;
  130.  
  131.     pd->pts += pd->frame_duration;
  132.  
  133.     return 0;
  134. }
  135.  
  136. static av_cold int pulse_close(AVFormatContext *s)
  137. {
  138.     PulseData *pd = s->priv_data;
  139.     pa_simple_free(pd->s);
  140.     return 0;
  141. }
  142.  
  143. #define OFFSET(a) offsetof(PulseData, a)
  144. #define D AV_OPT_FLAG_DECODING_PARAM
  145.  
  146. static const AVOption options[] = {
  147.     { "server",        "set PulseAudio server",                             OFFSET(server),        AV_OPT_TYPE_STRING, {.str = NULL},     0, 0, D },
  148.     { "name",          "set application name",                              OFFSET(name),          AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT},  0, 0, D },
  149.     { "stream_name",   "set stream description",                            OFFSET(stream_name),   AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
  150.     { "sample_rate",   "set sample rate in Hz",                             OFFSET(sample_rate),   AV_OPT_TYPE_INT,    {.i64 = 48000},    1, INT_MAX, D },
  151.     { "channels",      "set number of audio channels",                      OFFSET(channels),      AV_OPT_TYPE_INT,    {.i64 = 2},        1, INT_MAX, D },
  152.     { "frame_size",    "set number of bytes per frame",                     OFFSET(frame_size),    AV_OPT_TYPE_INT,    {.i64 = 1024},     1, INT_MAX, D },
  153.     { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT,    {.i64 = -1},      -1, INT_MAX, D },
  154.     { NULL },
  155. };
  156.  
  157. static const AVClass pulse_demuxer_class = {
  158.     .class_name     = "Pulse demuxer",
  159.     .item_name      = av_default_item_name,
  160.     .option         = options,
  161.     .version        = LIBAVUTIL_VERSION_INT,
  162. };
  163.  
  164. AVInputFormat ff_pulse_demuxer = {
  165.     .name           = "pulse",
  166.     .long_name      = NULL_IF_CONFIG_SMALL("Pulse audio input"),
  167.     .priv_data_size = sizeof(PulseData),
  168.     .read_header    = pulse_read_header,
  169.     .read_packet    = pulse_read_packet,
  170.     .read_close     = pulse_close,
  171.     .flags          = AVFMT_NOFILE,
  172.     .priv_class     = &pulse_demuxer_class,
  173. };
  174.