FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavdevice/alsa_enc.c
Date: 2022-12-05 03:11:11
Exec Total Coverage
Lines: 0 61 0.0%
Functions: 0 5 0.0%
Branches: 0 28 0.0%

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/internal.h"
43 #include "libavutil/time.h"
44
45
46 #include "libavformat/internal.h"
47 #include "libavformat/mux.h"
48 #include "avdevice.h"
49 #include "alsa.h"
50
51 static av_cold int audio_write_header(AVFormatContext *s1)
52 {
53 AlsaData *s = s1->priv_data;
54 AVStream *st = NULL;
55 unsigned int sample_rate;
56 enum AVCodecID codec_id;
57 int res;
58
59 if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
60 av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
61 return AVERROR(EINVAL);
62 }
63 st = s1->streams[0];
64
65 sample_rate = st->codecpar->sample_rate;
66 codec_id = st->codecpar->codec_id;
67 res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
68 st->codecpar->ch_layout.nb_channels, &codec_id);
69 if (sample_rate != st->codecpar->sample_rate) {
70 av_log(s1, AV_LOG_ERROR,
71 "sample rate %d not available, nearest is %d\n",
72 st->codecpar->sample_rate, sample_rate);
73 goto fail;
74 }
75 avpriv_set_pts_info(st, 64, 1, sample_rate);
76
77 return res;
78
79 fail:
80 snd_pcm_close(s->h);
81 return AVERROR(EIO);
82 }
83
84 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
85 {
86 AlsaData *s = s1->priv_data;
87 int res;
88 int size = pkt->size;
89 const uint8_t *buf = pkt->data;
90
91 size /= s->frame_size;
92 if (pkt->dts != AV_NOPTS_VALUE)
93 s->timestamp = pkt->dts;
94 s->timestamp += pkt->duration ? pkt->duration : size;
95
96 if (s->reorder_func) {
97 if (size > s->reorder_buf_size)
98 if (ff_alsa_extend_reorder_buf(s, size))
99 return AVERROR(ENOMEM);
100 s->reorder_func(buf, s->reorder_buf, size);
101 buf = s->reorder_buf;
102 }
103 while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
104 if (res == -EAGAIN) {
105
106 return AVERROR(EAGAIN);
107 }
108
109 if (ff_alsa_xrun_recover(s1, res) < 0) {
110 av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
111 snd_strerror(res));
112
113 return AVERROR(EIO);
114 }
115 }
116
117 return 0;
118 }
119
120 static int audio_write_frame(AVFormatContext *s1, int stream_index,
121 AVFrame **frame, unsigned flags)
122 {
123 AlsaData *s = s1->priv_data;
124 AVPacket pkt;
125
126 /* ff_alsa_open() should have accepted only supported formats */
127 if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
128 return av_sample_fmt_is_planar(s1->streams[stream_index]->codecpar->format) ?
129 AVERROR(EINVAL) : 0;
130 /* set only used fields */
131 pkt.data = (*frame)->data[0];
132 pkt.size = (*frame)->nb_samples * s->frame_size;
133 pkt.dts = (*frame)->pkt_dts;
134 #if FF_API_PKT_DURATION
135 FF_DISABLE_DEPRECATION_WARNINGS
136 if ((*frame)->pkt_duration)
137 pkt.duration = (*frame)->pkt_duration;
138 else
139 FF_ENABLE_DEPRECATION_WARNINGS
140 #endif
141 pkt.duration = (*frame)->duration;
142 return audio_write_packet(s1, &pkt);
143 }
144
145 static void
146 audio_get_output_timestamp(AVFormatContext *s1, int stream,
147 int64_t *dts, int64_t *wall)
148 {
149 AlsaData *s = s1->priv_data;
150 snd_pcm_sframes_t delay = 0;
151 *wall = av_gettime();
152 snd_pcm_delay(s->h, &delay);
153 *dts = s->timestamp - delay;
154 }
155
156 static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
157 {
158 return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
159 }
160
161 static const AVClass alsa_muxer_class = {
162 .class_name = "ALSA outdev",
163 .item_name = av_default_item_name,
164 .version = LIBAVUTIL_VERSION_INT,
165 .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
166 };
167
168 const AVOutputFormat ff_alsa_muxer = {
169 .name = "alsa",
170 .long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
171 .priv_data_size = sizeof(AlsaData),
172 .audio_codec = DEFAULT_CODEC_ID,
173 .video_codec = AV_CODEC_ID_NONE,
174 .write_header = audio_write_header,
175 .write_packet = audio_write_packet,
176 .write_trailer = ff_alsa_close,
177 .write_uncoded_frame = audio_write_frame,
178 .get_device_list = audio_get_device_list,
179 .get_output_timestamp = audio_get_output_timestamp,
180 .flags = AVFMT_NOFILE,
181 .priv_class = &alsa_muxer_class,
182 };
183