| 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: output | ||
| 26 | * @author Luca Abeni ( lucabe72 email it ) | ||
| 27 | * @author Benoit Fouet ( benoit fouet free fr ) | ||
| 28 | * | ||
| 29 | * This avdevice encoder can play audio to an ALSA (Advanced Linux | ||
| 30 | * Sound Architecture) device. | ||
| 31 | * | ||
| 32 | * The filename parameter is the name of an ALSA PCM device capable of | ||
| 33 | * capture, for example "default" or "plughw:1"; see the ALSA documentation | ||
| 34 | * for naming conventions. The empty string is equivalent to "default". | ||
| 35 | * | ||
| 36 | * The playback period is set to the lower value available for the device, | ||
| 37 | * which gives a low latency suitable for real-time playback. | ||
| 38 | */ | ||
| 39 | |||
| 40 | #include <alsa/asoundlib.h> | ||
| 41 | |||
| 42 | #include "libavutil/frame.h" | ||
| 43 | #include "libavutil/internal.h" | ||
| 44 | #include "libavutil/time.h" | ||
| 45 | |||
| 46 | |||
| 47 | #include "libavformat/internal.h" | ||
| 48 | #include "libavformat/mux.h" | ||
| 49 | #include "avdevice.h" | ||
| 50 | #include "alsa.h" | ||
| 51 | |||
| 52 | ✗ | static av_cold int audio_write_header(AVFormatContext *s1) | |
| 53 | { | ||
| 54 | ✗ | AlsaData *s = s1->priv_data; | |
| 55 | ✗ | AVStream *st = NULL; | |
| 56 | unsigned int sample_rate; | ||
| 57 | enum AVCodecID codec_id; | ||
| 58 | int res; | ||
| 59 | |||
| 60 | ✗ | if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) { | |
| 61 | ✗ | av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n"); | |
| 62 | ✗ | return AVERROR(EINVAL); | |
| 63 | } | ||
| 64 | ✗ | st = s1->streams[0]; | |
| 65 | |||
| 66 | ✗ | sample_rate = st->codecpar->sample_rate; | |
| 67 | ✗ | codec_id = st->codecpar->codec_id; | |
| 68 | ✗ | res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate, | |
| 69 | ✗ | st->codecpar->ch_layout.nb_channels, &codec_id); | |
| 70 | ✗ | if (sample_rate != st->codecpar->sample_rate) { | |
| 71 | ✗ | av_log(s1, AV_LOG_ERROR, | |
| 72 | "sample rate %d not available, nearest is %d\n", | ||
| 73 | ✗ | st->codecpar->sample_rate, sample_rate); | |
| 74 | ✗ | goto fail; | |
| 75 | } | ||
| 76 | ✗ | avpriv_set_pts_info(st, 64, 1, sample_rate); | |
| 77 | |||
| 78 | ✗ | return res; | |
| 79 | |||
| 80 | ✗ | fail: | |
| 81 | ✗ | snd_pcm_close(s->h); | |
| 82 | ✗ | return AVERROR(EIO); | |
| 83 | } | ||
| 84 | |||
| 85 | ✗ | static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) | |
| 86 | { | ||
| 87 | ✗ | AlsaData *s = s1->priv_data; | |
| 88 | int res; | ||
| 89 | ✗ | int size = pkt->size; | |
| 90 | ✗ | const uint8_t *buf = pkt->data; | |
| 91 | |||
| 92 | ✗ | size /= s->frame_size; | |
| 93 | ✗ | if (pkt->dts != AV_NOPTS_VALUE) | |
| 94 | ✗ | s->timestamp = pkt->dts; | |
| 95 | ✗ | s->timestamp += pkt->duration ? pkt->duration : size; | |
| 96 | |||
| 97 | ✗ | if (s->reorder_func) { | |
| 98 | ✗ | if (size > s->reorder_buf_size) | |
| 99 | ✗ | if (ff_alsa_extend_reorder_buf(s, size)) | |
| 100 | ✗ | return AVERROR(ENOMEM); | |
| 101 | ✗ | s->reorder_func(buf, s->reorder_buf, size); | |
| 102 | ✗ | buf = s->reorder_buf; | |
| 103 | } | ||
| 104 | ✗ | while ((res = snd_pcm_writei(s->h, buf, size)) < 0) { | |
| 105 | ✗ | if (res == -EAGAIN) { | |
| 106 | |||
| 107 | ✗ | return AVERROR(EAGAIN); | |
| 108 | } | ||
| 109 | |||
| 110 | ✗ | if (ff_alsa_xrun_recover(s1, res) < 0) { | |
| 111 | ✗ | av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n", | |
| 112 | snd_strerror(res)); | ||
| 113 | |||
| 114 | ✗ | return AVERROR(EIO); | |
| 115 | } | ||
| 116 | } | ||
| 117 | |||
| 118 | ✗ | return 0; | |
| 119 | } | ||
| 120 | |||
| 121 | ✗ | static int audio_write_frame(AVFormatContext *s1, int stream_index, | |
| 122 | AVFrame **frame, unsigned flags) | ||
| 123 | { | ||
| 124 | ✗ | AlsaData *s = s1->priv_data; | |
| 125 | AVPacket pkt; | ||
| 126 | |||
| 127 | /* ff_alsa_open() should have accepted only supported formats */ | ||
| 128 | ✗ | if ((flags & AV_WRITE_UNCODED_FRAME_QUERY)) | |
| 129 | ✗ | return av_sample_fmt_is_planar(s1->streams[stream_index]->codecpar->format) ? | |
| 130 | ✗ | AVERROR(EINVAL) : 0; | |
| 131 | /* set only used fields */ | ||
| 132 | ✗ | pkt.data = (*frame)->data[0]; | |
| 133 | ✗ | pkt.size = (*frame)->nb_samples * s->frame_size; | |
| 134 | ✗ | pkt.dts = (*frame)->pkt_dts; | |
| 135 | ✗ | pkt.duration = (*frame)->duration; | |
| 136 | ✗ | return audio_write_packet(s1, &pkt); | |
| 137 | } | ||
| 138 | |||
| 139 | static void | ||
| 140 | ✗ | audio_get_output_timestamp(AVFormatContext *s1, int stream, | |
| 141 | int64_t *dts, int64_t *wall) | ||
| 142 | { | ||
| 143 | ✗ | AlsaData *s = s1->priv_data; | |
| 144 | ✗ | snd_pcm_sframes_t delay = 0; | |
| 145 | ✗ | *wall = av_gettime(); | |
| 146 | ✗ | snd_pcm_delay(s->h, &delay); | |
| 147 | ✗ | *dts = s->timestamp - delay; | |
| 148 | ✗ | } | |
| 149 | |||
| 150 | ✗ | static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list) | |
| 151 | { | ||
| 152 | ✗ | return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK); | |
| 153 | } | ||
| 154 | |||
| 155 | static const AVClass alsa_muxer_class = { | ||
| 156 | .class_name = "ALSA outdev", | ||
| 157 | .item_name = av_default_item_name, | ||
| 158 | .version = LIBAVUTIL_VERSION_INT, | ||
| 159 | .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT, | ||
| 160 | }; | ||
| 161 | |||
| 162 | const FFOutputFormat ff_alsa_muxer = { | ||
| 163 | .p.name = "alsa", | ||
| 164 | .p.long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"), | ||
| 165 | .priv_data_size = sizeof(AlsaData), | ||
| 166 | .p.audio_codec = DEFAULT_CODEC_ID, | ||
| 167 | .p.video_codec = AV_CODEC_ID_NONE, | ||
| 168 | .write_header = audio_write_header, | ||
| 169 | .write_packet = audio_write_packet, | ||
| 170 | .write_trailer = ff_alsa_close, | ||
| 171 | .write_uncoded_frame = audio_write_frame, | ||
| 172 | .get_device_list = audio_get_device_list, | ||
| 173 | .get_output_timestamp = audio_get_output_timestamp, | ||
| 174 | .p.flags = AVFMT_NOFILE, | ||
| 175 | .p.priv_class = &alsa_muxer_class, | ||
| 176 | }; | ||
| 177 |