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* ALSA input and output
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* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
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* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
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* This file is part of FFmpeg.
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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* @file libavdevice/alsa-audio-dec.c
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* ALSA input and output: input
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* @author Luca Abeni ( lucabe72 email it )
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* @author Benoit Fouet ( benoit fouet free fr )
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* @author Nicolas George ( nicolas george normalesup org )
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* This avdevice decoder allows to capture audio from an ALSA (Advanced
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* Linux Sound Architecture) device.
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* The filename parameter is the name of an ALSA PCM device capable of
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* capture, for example "default" or "plughw:1"; see the ALSA documentation
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* for naming conventions. The empty string is equivalent to "default".
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* The capture period is set to the lower value available for the device,
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* which gives a low latency suitable for real-time capture.
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* The PTS are an Unix time in microsecond.
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* Due to a bug in the ALSA library
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* (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
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* decoder does not work with certain ALSA plugins, especially the dsnoop
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#include "libavformat/avformat.h"
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#include <alsa/asoundlib.h>
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#include "alsa-audio.h"
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av_cold static int audio_read_header(AVFormatContext *s1,
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AVFormatParameters *ap)
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AlsaData *s = s1->priv_data;
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unsigned int sample_rate;
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snd_pcm_sw_params_t *sw_params;
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if (ap->sample_rate <= 0) {
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av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate);
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if (ap->channels <= 0) {
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av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels);
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st = av_new_stream(s1, 0);
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av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
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return AVERROR(ENOMEM);
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sample_rate = ap->sample_rate;
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codec_id = ap->audio_codec_id;
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ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels,
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if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
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av_log(s1, AV_LOG_WARNING,
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"capture with some ALSA plugins, especially dsnoop, "
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ret = snd_pcm_sw_params_malloc(&sw_params);
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av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
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snd_pcm_sw_params_current(s->h, sw_params);
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snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
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ret = snd_pcm_sw_params(s->h, sw_params);
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snd_pcm_sw_params_free(sw_params);
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av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
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/* take real parameters */
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st->codec->codec_type = CODEC_TYPE_AUDIO;
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st->codec->codec_id = codec_id;
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st->codec->sample_rate = sample_rate;
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st->codec->channels = ap->channels;
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av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
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AlsaData *s = s1->priv_data;
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AVStream *st = s1->streams[0];
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snd_htimestamp_t timestamp;
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snd_pcm_uframes_t ts_delay;
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if (av_new_packet(pkt, s->period_size) < 0) {
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while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
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if (res == -EAGAIN) {
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return AVERROR(EAGAIN);
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if (ff_alsa_xrun_recover(s1, res) < 0) {
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av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
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snd_pcm_htimestamp(s->h, &ts_delay, ×tamp);
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pkt->pts = timestamp.tv_sec * 1000000LL
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+ (timestamp.tv_nsec * st->codec->sample_rate
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- ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
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/ (st->codec->sample_rate * 1000LL);
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pkt->size = res * s->frame_size;
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AVInputFormat alsa_demuxer = {
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NULL_IF_CONFIG_SMALL("ALSA audio input"),
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.flags = AVFMT_NOFILE,