| Line | Branch | Exec | Source |
|---|---|---|---|
| 1 | /* | ||
| 2 | * ALSA input and output | ||
| 3 | * Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) | ||
| 4 | * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) | ||
| 5 | * | ||
| 6 | * This file is part of FFmpeg. | ||
| 7 | * | ||
| 8 | * FFmpeg is free software; you can redistribute it and/or | ||
| 9 | * modify it under the terms of the GNU Lesser General Public | ||
| 10 | * License as published by the Free Software Foundation; either | ||
| 11 | * version 2.1 of the License, or (at your option) any later version. | ||
| 12 | * | ||
| 13 | * FFmpeg is distributed in the hope that it will be useful, | ||
| 14 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | ||
| 15 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | ||
| 16 | * Lesser General Public License for more details. | ||
| 17 | * | ||
| 18 | * You should have received a copy of the GNU Lesser General Public | ||
| 19 | * License along with FFmpeg; if not, write to the Free Software | ||
| 20 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | ||
| 21 | */ | ||
| 22 | |||
| 23 | /** | ||
| 24 | * @file | ||
| 25 | * ALSA input and output: input | ||
| 26 | * @author Luca Abeni ( lucabe72 email it ) | ||
| 27 | * @author Benoit Fouet ( benoit fouet free fr ) | ||
| 28 | * @author Nicolas George ( nicolas george normalesup org ) | ||
| 29 | * | ||
| 30 | * This avdevice decoder can capture audio from an ALSA (Advanced | ||
| 31 | * Linux Sound Architecture) device. | ||
| 32 | * | ||
| 33 | * The filename parameter is the name of an ALSA PCM device capable of | ||
| 34 | * capture, for example "default" or "plughw:1"; see the ALSA documentation | ||
| 35 | * for naming conventions. The empty string is equivalent to "default". | ||
| 36 | * | ||
| 37 | * The capture period is set to the lower value available for the device, | ||
| 38 | * which gives a low latency suitable for real-time capture. | ||
| 39 | * | ||
| 40 | * The PTS are an Unix time in microsecond. | ||
| 41 | * | ||
| 42 | * Due to a bug in the ALSA library | ||
| 43 | * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this | ||
| 44 | * decoder does not work with certain ALSA plugins, especially the dsnoop | ||
| 45 | * plugin. | ||
| 46 | */ | ||
| 47 | |||
| 48 | #include <alsa/asoundlib.h> | ||
| 49 | |||
| 50 | #include "libavutil/internal.h" | ||
| 51 | #include "libavutil/mathematics.h" | ||
| 52 | #include "libavutil/opt.h" | ||
| 53 | #include "libavutil/time.h" | ||
| 54 | |||
| 55 | #include "libavformat/demux.h" | ||
| 56 | #include "libavformat/internal.h" | ||
| 57 | |||
| 58 | #include "avdevice.h" | ||
| 59 | #include "alsa.h" | ||
| 60 | |||
| 61 | ✗ | static av_cold int audio_read_header(AVFormatContext *s1) | |
| 62 | { | ||
| 63 | ✗ | AlsaData *s = s1->priv_data; | |
| 64 | AVStream *st; | ||
| 65 | int ret; | ||
| 66 | enum AVCodecID codec_id; | ||
| 67 | |||
| 68 | ✗ | st = avformat_new_stream(s1, NULL); | |
| 69 | ✗ | if (!st) { | |
| 70 | ✗ | av_log(s1, AV_LOG_ERROR, "Cannot add stream\n"); | |
| 71 | |||
| 72 | ✗ | return AVERROR(ENOMEM); | |
| 73 | } | ||
| 74 | ✗ | codec_id = s1->audio_codec_id; | |
| 75 | |||
| 76 | #if FF_API_ALSA_CHANNELS | ||
| 77 | ✗ | if (s->channels > 0) { | |
| 78 | ✗ | av_channel_layout_uninit(&s->ch_layout); | |
| 79 | ✗ | s->ch_layout.nb_channels = s->channels; | |
| 80 | } | ||
| 81 | #endif | ||
| 82 | |||
| 83 | ✗ | ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, &s->ch_layout, | |
| 84 | &codec_id); | ||
| 85 | ✗ | if (ret < 0) { | |
| 86 | ✗ | return AVERROR(EIO); | |
| 87 | } | ||
| 88 | |||
| 89 | /* take real parameters */ | ||
| 90 | ✗ | st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; | |
| 91 | ✗ | st->codecpar->codec_id = codec_id; | |
| 92 | ✗ | st->codecpar->sample_rate = s->sample_rate; | |
| 93 | ✗ | ret = av_channel_layout_copy(&st->codecpar->ch_layout, &s->ch_layout); | |
| 94 | ✗ | if (ret < 0) | |
| 95 | ✗ | goto fail; | |
| 96 | ✗ | st->codecpar->frame_size = s->frame_size; | |
| 97 | ✗ | avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ | |
| 98 | /* microseconds instead of seconds, MHz instead of Hz */ | ||
| 99 | ✗ | s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate, | |
| 100 | ✗ | s->period_size, 1.5E-6); | |
| 101 | ✗ | if (!s->timefilter) { | |
| 102 | ✗ | ret = AVERROR(EIO); | |
| 103 | ✗ | goto fail; | |
| 104 | } | ||
| 105 | |||
| 106 | ✗ | return 0; | |
| 107 | |||
| 108 | ✗ | fail: | |
| 109 | ✗ | snd_pcm_close(s->h); | |
| 110 | ✗ | return ret; | |
| 111 | } | ||
| 112 | |||
| 113 | ✗ | static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) | |
| 114 | { | ||
| 115 | ✗ | AlsaData *s = s1->priv_data; | |
| 116 | int res; | ||
| 117 | int64_t dts; | ||
| 118 | ✗ | snd_pcm_sframes_t delay = 0; | |
| 119 | |||
| 120 | ✗ | if (!s->pkt->data) { | |
| 121 | ✗ | int ret = av_new_packet(s->pkt, s->period_size * s->frame_size); | |
| 122 | ✗ | if (ret < 0) | |
| 123 | ✗ | return ret; | |
| 124 | ✗ | s->pkt->size = 0; | |
| 125 | } | ||
| 126 | |||
| 127 | do { | ||
| 128 | ✗ | while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, s->period_size - s->pkt->size / s->frame_size)) < 0) { | |
| 129 | ✗ | if (res == -EAGAIN) { | |
| 130 | ✗ | return AVERROR(EAGAIN); | |
| 131 | } | ||
| 132 | ✗ | s->pkt->size = 0; | |
| 133 | ✗ | if (ff_alsa_xrun_recover(s1, res) < 0) { | |
| 134 | ✗ | av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n", | |
| 135 | snd_strerror(res)); | ||
| 136 | ✗ | return AVERROR(EIO); | |
| 137 | } | ||
| 138 | ✗ | ff_timefilter_reset(s->timefilter); | |
| 139 | } | ||
| 140 | ✗ | s->pkt->size += res * s->frame_size; | |
| 141 | ✗ | } while (s->pkt->size < s->period_size * s->frame_size); | |
| 142 | |||
| 143 | ✗ | av_packet_move_ref(pkt, s->pkt); | |
| 144 | ✗ | dts = av_gettime(); | |
| 145 | ✗ | snd_pcm_delay(s->h, &delay); | |
| 146 | ✗ | dts -= av_rescale(delay + res, 1000000, s->sample_rate); | |
| 147 | ✗ | pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period); | |
| 148 | ✗ | s->last_period = res; | |
| 149 | |||
| 150 | ✗ | return 0; | |
| 151 | } | ||
| 152 | |||
| 153 | ✗ | static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list) | |
| 154 | { | ||
| 155 | ✗ | return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE); | |
| 156 | } | ||
| 157 | |||
| 158 | static const AVOption options[] = { | ||
| 159 | { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, | ||
| 160 | #if FF_API_ALSA_CHANNELS | ||
| 161 | { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_DEPRECATED }, | ||
| 162 | #endif | ||
| 163 | { "ch_layout", "", offsetof(AlsaData, ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str = "2C"}, INT_MIN, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, | ||
| 164 | { NULL }, | ||
| 165 | }; | ||
| 166 | |||
| 167 | static const AVClass alsa_demuxer_class = { | ||
| 168 | .class_name = "ALSA indev", | ||
| 169 | .item_name = av_default_item_name, | ||
| 170 | .option = options, | ||
| 171 | .version = LIBAVUTIL_VERSION_INT, | ||
| 172 | .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT, | ||
| 173 | }; | ||
| 174 | |||
| 175 | const FFInputFormat ff_alsa_demuxer = { | ||
| 176 | .p.name = "alsa", | ||
| 177 | .p.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"), | ||
| 178 | .p.flags = AVFMT_NOFILE, | ||
| 179 | .p.priv_class = &alsa_demuxer_class, | ||
| 180 | .priv_data_size = sizeof(AlsaData), | ||
| 181 | .read_header = audio_read_header, | ||
| 182 | .read_packet = audio_read_packet, | ||
| 183 | .read_close = ff_alsa_close, | ||
| 184 | .get_device_list = audio_get_device_list, | ||
| 185 | }; | ||
| 186 |