| /* |
| * 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 (av_new_packet(pkt, s->period_size * s->frame_size) < 0) { |
| return AVERROR(EIO); |
| } |
| |
| while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) { |
| if (res == -EAGAIN) { |
| av_packet_unref(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_packet_unref(pkt); |
| |
| return AVERROR(EIO); |
| } |
| ff_timefilter_reset(s->timefilter); |
| } |
| |
| 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; |
| |
| pkt->size = res * s->frame_size; |
| |
| 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 demuxer", |
| .item_name = av_default_item_name, |
| .option = options, |
| .version = LIBAVUTIL_VERSION_INT, |
| .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT, |
| }; |
| |
| 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, |
| }; |