alsa_enc.c 5.4 KB

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  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: output
  25. * @author Luca Abeni ( lucabe72 email it )
  26. * @author Benoit Fouet ( benoit fouet free fr )
  27. *
  28. * This avdevice encoder can play audio to an ALSA (Advanced Linux
  29. * Sound Architecture) device.
  30. *
  31. * The filename parameter is the name of an ALSA PCM device capable of
  32. * capture, for example "default" or "plughw:1"; see the ALSA documentation
  33. * for naming conventions. The empty string is equivalent to "default".
  34. *
  35. * The playback period is set to the lower value available for the device,
  36. * which gives a low latency suitable for real-time playback.
  37. */
  38. #include <alsa/asoundlib.h>
  39. #include "libavutil/internal.h"
  40. #include "libavutil/time.h"
  41. #include "libavformat/internal.h"
  42. #include "avdevice.h"
  43. #include "alsa.h"
  44. static av_cold int audio_write_header(AVFormatContext *s1)
  45. {
  46. AlsaData *s = s1->priv_data;
  47. AVStream *st = NULL;
  48. unsigned int sample_rate;
  49. enum AVCodecID codec_id;
  50. int res;
  51. if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
  52. av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
  53. return AVERROR(EINVAL);
  54. }
  55. st = s1->streams[0];
  56. sample_rate = st->codecpar->sample_rate;
  57. codec_id = st->codecpar->codec_id;
  58. res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
  59. st->codecpar->channels, &codec_id);
  60. if (sample_rate != st->codecpar->sample_rate) {
  61. av_log(s1, AV_LOG_ERROR,
  62. "sample rate %d not available, nearest is %d\n",
  63. st->codecpar->sample_rate, sample_rate);
  64. goto fail;
  65. }
  66. avpriv_set_pts_info(st, 64, 1, sample_rate);
  67. return res;
  68. fail:
  69. snd_pcm_close(s->h);
  70. return AVERROR(EIO);
  71. }
  72. static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
  73. {
  74. AlsaData *s = s1->priv_data;
  75. int res;
  76. int size = pkt->size;
  77. uint8_t *buf = pkt->data;
  78. size /= s->frame_size;
  79. if (pkt->dts != AV_NOPTS_VALUE)
  80. s->timestamp = pkt->dts;
  81. s->timestamp += pkt->duration ? pkt->duration : size;
  82. if (s->reorder_func) {
  83. if (size > s->reorder_buf_size)
  84. if (ff_alsa_extend_reorder_buf(s, size))
  85. return AVERROR(ENOMEM);
  86. s->reorder_func(buf, s->reorder_buf, size);
  87. buf = s->reorder_buf;
  88. }
  89. while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
  90. if (res == -EAGAIN) {
  91. return AVERROR(EAGAIN);
  92. }
  93. if (ff_alsa_xrun_recover(s1, res) < 0) {
  94. av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
  95. snd_strerror(res));
  96. return AVERROR(EIO);
  97. }
  98. }
  99. return 0;
  100. }
  101. static int audio_write_frame(AVFormatContext *s1, int stream_index,
  102. AVFrame **frame, unsigned flags)
  103. {
  104. AlsaData *s = s1->priv_data;
  105. AVPacket pkt;
  106. /* ff_alsa_open() should have accepted only supported formats */
  107. if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
  108. return av_sample_fmt_is_planar(s1->streams[stream_index]->codecpar->format) ?
  109. AVERROR(EINVAL) : 0;
  110. /* set only used fields */
  111. pkt.data = (*frame)->data[0];
  112. pkt.size = (*frame)->nb_samples * s->frame_size;
  113. pkt.dts = (*frame)->pkt_dts;
  114. pkt.duration = av_frame_get_pkt_duration(*frame);
  115. return audio_write_packet(s1, &pkt);
  116. }
  117. static void
  118. audio_get_output_timestamp(AVFormatContext *s1, int stream,
  119. int64_t *dts, int64_t *wall)
  120. {
  121. AlsaData *s = s1->priv_data;
  122. snd_pcm_sframes_t delay = 0;
  123. *wall = av_gettime();
  124. snd_pcm_delay(s->h, &delay);
  125. *dts = s->timestamp - delay;
  126. }
  127. static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
  128. {
  129. return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
  130. }
  131. static const AVClass alsa_muxer_class = {
  132. .class_name = "ALSA muxer",
  133. .item_name = av_default_item_name,
  134. .version = LIBAVUTIL_VERSION_INT,
  135. .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
  136. };
  137. AVOutputFormat ff_alsa_muxer = {
  138. .name = "alsa",
  139. .long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
  140. .priv_data_size = sizeof(AlsaData),
  141. .audio_codec = DEFAULT_CODEC_ID,
  142. .video_codec = AV_CODEC_ID_NONE,
  143. .write_header = audio_write_header,
  144. .write_packet = audio_write_packet,
  145. .write_trailer = ff_alsa_close,
  146. .write_uncoded_frame = audio_write_frame,
  147. .get_device_list = audio_get_device_list,
  148. .get_output_timestamp = audio_get_output_timestamp,
  149. .flags = AVFMT_NOFILE,
  150. .priv_class = &alsa_muxer_class,
  151. };