/* * ALSA input and output * Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) * * 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 * ALSA input and output: input * @author Luca Abeni ( lucabe72 email it ) * @author Benoit Fouet ( benoit fouet free fr ) * @author Nicolas George ( nicolas george normalesup org ) * * This avdevice decoder can capture audio from an ALSA (Advanced * Linux Sound Architecture) device. * * The filename parameter is the name of an ALSA PCM device capable of * capture, for example "default" or "plughw:1"; see the ALSA documentation * for naming conventions. The empty string is equivalent to "default". * * The capture period is set to the lower value available for the device, * which gives a low latency suitable for real-time capture. * * The PTS are an Unix time in microsecond. * * Due to a bug in the ALSA library * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this * decoder does not work with certain ALSA plugins, especially the dsnoop * plugin. */ #include <alsa/asoundlib.h> #include "libavutil/internal.h" #include "libavutil/mathematics.h" #include "libavutil/opt.h" #include "libavutil/time.h" #include "libavformat/internal.h" #include "avdevice.h" #include "alsa.h" static av_cold int audio_read_header(AVFormatContext *s1) { AlsaData *s = s1->priv_data; AVStream *st; int ret; enum AVCodecID codec_id; st = avformat_new_stream(s1, NULL); if (!st) { av_log(s1, AV_LOG_ERROR, "Cannot add stream\n"); return AVERROR(ENOMEM); } codec_id = s1->audio_codec_id; ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels, &codec_id); if (ret < 0) { return AVERROR(EIO); } /* take real parameters */ st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; st->codecpar->codec_id = codec_id; st->codecpar->sample_rate = s->sample_rate; st->codecpar->channels = s->channels; st->codecpar->frame_size = s->frame_size; avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ /* microseconds instead of seconds, MHz instead of Hz */ s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate, s->period_size, 1.5E-6); if (!s->timefilter) goto fail; return 0; fail: snd_pcm_close(s->h); return AVERROR(EIO); } static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) { AlsaData *s = s1->priv_data; int res; int64_t dts; snd_pcm_sframes_t delay = 0; if (!s->pkt->data) { int ret = av_new_packet(s->pkt, s->period_size * s->frame_size); if (ret < 0) return ret; s->pkt->size = 0; } do { while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, s->period_size - s->pkt->size / s->frame_size)) < 0) { if (res == -EAGAIN) { return AVERROR(EAGAIN); } s->pkt->size = 0; if (ff_alsa_xrun_recover(s1, res) < 0) { av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n", snd_strerror(res)); return AVERROR(EIO); } ff_timefilter_reset(s->timefilter); } s->pkt->size += res * s->frame_size; } while (s->pkt->size < s->period_size * s->frame_size); av_packet_move_ref(pkt, s->pkt); dts = av_gettime(); snd_pcm_delay(s->h, &delay); dts -= av_rescale(delay + res, 1000000, s->sample_rate); pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period); s->last_period = res; return 0; } static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list) { return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE); } static const AVOption options[] = { { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, { NULL }, }; static const AVClass alsa_demuxer_class = { .class_name = "ALSA indev", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT, }; const AVInputFormat ff_alsa_demuxer = { .name = "alsa", .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"), .priv_data_size = sizeof(AlsaData), .read_header = audio_read_header, .read_packet = audio_read_packet, .read_close = ff_alsa_close, .get_device_list = audio_get_device_list, .flags = AVFMT_NOFILE, .priv_class = &alsa_demuxer_class, };