diff options
author | Diego Biurrun <diego@biurrun.de> | 2015-03-27 12:40:23 +0100 |
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committer | Diego Biurrun <diego@biurrun.de> | 2015-04-09 16:40:12 +0200 |
commit | 8d26c193fb42d08602ac93ece039d4718d029adc (patch) | |
tree | ced4ff3f94794ab9ca513dd67cedd4519ffeb591 /libavdevice/alsa-audio-dec.c | |
parent | c201069fac9a76e6604f9d84d76a172434d62200 (diff) |
avdevice: Apply a more consistent file naming scheme
Diffstat (limited to 'libavdevice/alsa-audio-dec.c')
-rw-r--r-- | libavdevice/alsa-audio-dec.c | 178 |
1 files changed, 0 insertions, 178 deletions
diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c deleted file mode 100644 index a6a814b80f..0000000000 --- a/libavdevice/alsa-audio-dec.c +++ /dev/null @@ -1,178 +0,0 @@ -/* - * 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-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, -}; |