| /* |
| * 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 libavdevice/alsa-audio-dec.c |
| * 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 "libavformat/avformat.h" |
| #include <alsa/asoundlib.h> |
| |
| #include "alsa-audio.h" |
| |
| av_cold static int audio_read_header(AVFormatContext *s1, |
| AVFormatParameters *ap) |
| { |
| AlsaData *s = s1->priv_data; |
| AVStream *st; |
| int ret; |
| unsigned int sample_rate; |
| int codec_id; |
| snd_pcm_sw_params_t *sw_params; |
| |
| if (ap->sample_rate <= 0) { |
| av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate); |
| |
| return AVERROR(EIO); |
| } |
| |
| if (ap->channels <= 0) { |
| av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels); |
| |
| return AVERROR(EIO); |
| } |
| |
| st = av_new_stream(s1, 0); |
| if (!st) { |
| av_log(s1, AV_LOG_ERROR, "Cannot add stream\n"); |
| |
| return AVERROR(ENOMEM); |
| } |
| sample_rate = ap->sample_rate; |
| codec_id = ap->audio_codec_id; |
| |
| ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->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 = CODEC_TYPE_AUDIO; |
| st->codec->codec_id = codec_id; |
| st->codec->sample_rate = sample_rate; |
| st->codec->channels = ap->channels; |
| av_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 |
| - ts_delay * 1000000000LL + st->codec->sample_rate * 500LL) |
| / (st->codec->sample_rate * 1000LL); |
| |
| pkt->size = res * s->frame_size; |
| |
| return 0; |
| } |
| |
| AVInputFormat alsa_demuxer = { |
| "alsa", |
| NULL_IF_CONFIG_SMALL("ALSA audio input"), |
| sizeof(AlsaData), |
| NULL, |
| audio_read_header, |
| audio_read_packet, |
| ff_alsa_close, |
| .flags = AVFMT_NOFILE, |
| }; |