alsa_dec.c 5.2 KB

123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101102103104105106107108109110111112113114115116117118119120121122123124125126127128129130131132133134135136137138139140141142143144145146147148149150151152153154155156157158159160161162163164165166167168
  1. /*
  2. * ALSA input and output
  3. * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
  4. * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
  5. *
  6. * This file is part of FFmpeg.
  7. *
  8. * FFmpeg is free software; you can redistribute it and/or
  9. * modify it under the terms of the GNU Lesser General Public
  10. * License as published by the Free Software Foundation; either
  11. * version 2.1 of the License, or (at your option) any later version.
  12. *
  13. * FFmpeg is distributed in the hope that it will be useful,
  14. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  16. * Lesser General Public License for more details.
  17. *
  18. * You should have received a copy of the GNU Lesser General Public
  19. * License along with FFmpeg; if not, write to the Free Software
  20. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21. */
  22. /**
  23. * @file
  24. * ALSA input and output: input
  25. * @author Luca Abeni ( lucabe72 email it )
  26. * @author Benoit Fouet ( benoit fouet free fr )
  27. * @author Nicolas George ( nicolas george normalesup org )
  28. *
  29. * This avdevice decoder can capture audio from an ALSA (Advanced
  30. * Linux Sound Architecture) device.
  31. *
  32. * The filename parameter is the name of an ALSA PCM device capable of
  33. * capture, for example "default" or "plughw:1"; see the ALSA documentation
  34. * for naming conventions. The empty string is equivalent to "default".
  35. *
  36. * The capture period is set to the lower value available for the device,
  37. * which gives a low latency suitable for real-time capture.
  38. *
  39. * The PTS are an Unix time in microsecond.
  40. *
  41. * Due to a bug in the ALSA library
  42. * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
  43. * decoder does not work with certain ALSA plugins, especially the dsnoop
  44. * plugin.
  45. */
  46. #include <alsa/asoundlib.h>
  47. #include "libavutil/internal.h"
  48. #include "libavutil/mathematics.h"
  49. #include "libavutil/opt.h"
  50. #include "libavutil/time.h"
  51. #include "libavformat/internal.h"
  52. #include "avdevice.h"
  53. #include "alsa.h"
  54. static av_cold int audio_read_header(AVFormatContext *s1)
  55. {
  56. AlsaData *s = s1->priv_data;
  57. AVStream *st;
  58. int ret;
  59. enum AVCodecID codec_id;
  60. st = avformat_new_stream(s1, NULL);
  61. if (!st) {
  62. av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
  63. return AVERROR(ENOMEM);
  64. }
  65. codec_id = s1->audio_codec_id;
  66. ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
  67. &codec_id);
  68. if (ret < 0) {
  69. return AVERROR(EIO);
  70. }
  71. /* take real parameters */
  72. st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
  73. st->codecpar->codec_id = codec_id;
  74. st->codecpar->sample_rate = s->sample_rate;
  75. st->codecpar->channels = s->channels;
  76. st->codecpar->frame_size = s->frame_size;
  77. avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
  78. /* microseconds instead of seconds, MHz instead of Hz */
  79. s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
  80. s->period_size, 1.5E-6);
  81. if (!s->timefilter)
  82. goto fail;
  83. return 0;
  84. fail:
  85. snd_pcm_close(s->h);
  86. return AVERROR(EIO);
  87. }
  88. static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
  89. {
  90. AlsaData *s = s1->priv_data;
  91. int res;
  92. int64_t dts;
  93. snd_pcm_sframes_t delay = 0;
  94. if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
  95. return AVERROR(EIO);
  96. }
  97. while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
  98. if (res == -EAGAIN) {
  99. av_packet_unref(pkt);
  100. return AVERROR(EAGAIN);
  101. }
  102. if (ff_alsa_xrun_recover(s1, res) < 0) {
  103. av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
  104. snd_strerror(res));
  105. av_packet_unref(pkt);
  106. return AVERROR(EIO);
  107. }
  108. ff_timefilter_reset(s->timefilter);
  109. }
  110. dts = av_gettime();
  111. snd_pcm_delay(s->h, &delay);
  112. dts -= av_rescale(delay + res, 1000000, s->sample_rate);
  113. pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
  114. s->last_period = res;
  115. pkt->size = res * s->frame_size;
  116. return 0;
  117. }
  118. static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
  119. {
  120. return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
  121. }
  122. static const AVOption options[] = {
  123. { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
  124. { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
  125. { NULL },
  126. };
  127. static const AVClass alsa_demuxer_class = {
  128. .class_name = "ALSA demuxer",
  129. .item_name = av_default_item_name,
  130. .option = options,
  131. .version = LIBAVUTIL_VERSION_INT,
  132. .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
  133. };
  134. AVInputFormat ff_alsa_demuxer = {
  135. .name = "alsa",
  136. .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
  137. .priv_data_size = sizeof(AlsaData),
  138. .read_header = audio_read_header,
  139. .read_packet = audio_read_packet,
  140. .read_close = ff_alsa_close,
  141. .get_device_list = audio_get_device_list,
  142. .flags = AVFMT_NOFILE,
  143. .priv_class = &alsa_demuxer_class,
  144. };