diff options
Diffstat (limited to 'libavdevice/alsa_dec.c')
| -rw-r--r-- | libavdevice/alsa_dec.c | 178 |
1 files changed, 178 insertions, 0 deletions
diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c new file mode 100644 index 0000000000..2cc5b7d574 --- /dev/null +++ b/libavdevice/alsa_dec.c @@ -0,0 +1,178 @@ +/* + * 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 "libavutil/internal.h" +#include "libavutil/opt.h" + +#include "libavformat/avformat.h" +#include "libavformat/internal.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; + 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, +}; |