/* * 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 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 * 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 allows to 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 "libavformat/avformat.h" #include "libavformat/internal.h" #include "libavutil/opt.h" #include "alsa-audio.h" static av_cold int audio_read_header(AVFormatContext *s1) { AlsaData *s = s1->priv_data; AVStream *st; int ret; enum AVCodecID codec_id; snd_pcm_sw_params_t *sw_params; 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); } if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW) av_log(s1, AV_LOG_WARNING, "capture with some ALSA plugins, especially dsnoop, " "may hang.\n"); ret = snd_pcm_sw_params_malloc(&sw_params); if (ret < 0) { av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n", snd_strerror(ret)); goto fail; } snd_pcm_sw_params_current(s->h, sw_params); snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE); ret = snd_pcm_sw_params(s->h, sw_params); snd_pcm_sw_params_free(sw_params); if (ret < 0) { av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n", snd_strerror(ret)); goto fail; } /* take real parameters */ st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_id = codec_id; st->codec->sample_rate = s->sample_rate; st->codec->channels = s->channels; avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ 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; AVStream *st = s1->streams[0]; int res; snd_htimestamp_t timestamp; snd_pcm_uframes_t ts_delay; if (av_new_packet(pkt, s->period_size) < 0) { return AVERROR(EIO); } while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) { if (res == -EAGAIN) { av_free_packet(pkt); return AVERROR(EAGAIN); } if (ff_alsa_xrun_recover(s1, res) < 0) { av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n", snd_strerror(res)); av_free_packet(pkt); return AVERROR(EIO); } } snd_pcm_htimestamp(s->h, &ts_delay, ×tamp); ts_delay += res; pkt->pts = timestamp.tv_sec * 1000000LL + (timestamp.tv_nsec * st->codec->sample_rate - (int64_t)ts_delay * 1000000000LL + st->codec->sample_rate * 500LL) / (st->codec->sample_rate * 1000LL); pkt->size = res * s->frame_size; return 0; } 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 demuxer", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; 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, .flags = AVFMT_NOFILE, .priv_class = &alsa_demuxer_class, };