FFmpeg  4.4.6
alsa_dec.c
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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/internal.h"
56 
57 #include "avdevice.h"
58 #include "alsa.h"
59 
61 {
62  AlsaData *s = s1->priv_data;
63  AVStream *st;
64  int ret;
65  enum AVCodecID codec_id;
66 
68  if (!st) {
69  av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
70 
71  return AVERROR(ENOMEM);
72  }
73  codec_id = s1->audio_codec_id;
74 
75  ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
76  &codec_id);
77  if (ret < 0) {
78  return AVERROR(EIO);
79  }
80 
81  /* take real parameters */
83  st->codecpar->codec_id = codec_id;
84  st->codecpar->sample_rate = s->sample_rate;
85  st->codecpar->channels = s->channels;
86  st->codecpar->frame_size = s->frame_size;
87  avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
88  /* microseconds instead of seconds, MHz instead of Hz */
89  s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
90  s->period_size, 1.5E-6);
91  if (!s->timefilter)
92  goto fail;
93 
94  return 0;
95 
96 fail:
97  snd_pcm_close(s->h);
98  return AVERROR(EIO);
99 }
100 
102 {
103  AlsaData *s = s1->priv_data;
104  int res;
105  int64_t dts;
106  snd_pcm_sframes_t delay = 0;
107 
108  if (!s->pkt->data) {
109  int ret = av_new_packet(s->pkt, s->period_size * s->frame_size);
110  if (ret < 0)
111  return ret;
112  s->pkt->size = 0;
113  }
114 
115  do {
116  while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, s->period_size - s->pkt->size / s->frame_size)) < 0) {
117  if (res == -EAGAIN) {
118  return AVERROR(EAGAIN);
119  }
120  s->pkt->size = 0;
121  if (ff_alsa_xrun_recover(s1, res) < 0) {
122  av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
123  snd_strerror(res));
124  return AVERROR(EIO);
125  }
126  ff_timefilter_reset(s->timefilter);
127  }
128  s->pkt->size += res * s->frame_size;
129  } while (s->pkt->size < s->period_size * s->frame_size);
130 
131  av_packet_move_ref(pkt, s->pkt);
132  dts = av_gettime();
133  snd_pcm_delay(s->h, &delay);
134  dts -= av_rescale(delay + res, 1000000, s->sample_rate);
135  pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
136  s->last_period = res;
137 
138  return 0;
139 }
140 
142 {
143  return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
144 }
145 
146 static const AVOption options[] = {
147  { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
148  { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
149  { NULL },
150 };
151 
152 static const AVClass alsa_demuxer_class = {
153  .class_name = "ALSA indev",
154  .item_name = av_default_item_name,
155  .option = options,
156  .version = LIBAVUTIL_VERSION_INT,
158 };
159 
161  .name = "alsa",
162  .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
163  .priv_data_size = sizeof(AlsaData),
167  .get_device_list = audio_get_device_list,
168  .flags = AVFMT_NOFILE,
169  .priv_class = &alsa_demuxer_class,
170 };
int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
Try to recover from ALSA buffer underrun.
Definition: alsa.c:319
int ff_alsa_get_device_list(AVDeviceInfoList *device_list, snd_pcm_stream_t stream_type)
Definition: alsa.c:357
av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode, unsigned int *sample_rate, int channels, enum AVCodecID *codec_id)
Open an ALSA PCM.
Definition: alsa.c:167
av_cold int ff_alsa_close(AVFormatContext *s1)
Close the ALSA PCM.
Definition: alsa.c:303
ALSA input and output: definitions and structures.
static av_cold int audio_read_header(AVFormatContext *s1)
Definition: alsa_dec.c:60
static const AVOption options[]
Definition: alsa_dec.c:146
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: alsa_dec.c:101
AVInputFormat ff_alsa_demuxer
Definition: alsa_dec.c:160
static const AVClass alsa_demuxer_class
Definition: alsa_dec.c:152
static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
Definition: alsa_dec.c:141
channels
Definition: aptx.h:33
#define av_cold
Definition: attributes.h:88
Main libavdevice API header.
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:458
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
#define flags(name, subs,...)
Definition: cbs_av1.c:572
#define s(width, name)
Definition: cbs_vp9.c:257
#define fail()
Definition: checkasm.h:133
#define NULL
Definition: coverity.c:32
long long int64_t
Definition: coverity.c:34
sample_rate
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:545
@ AV_OPT_TYPE_INT
Definition: opt.h:225
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: codec_id.h:46
void av_packet_move_ref(AVPacket *dst, AVPacket *src)
Move every field in src to dst and reset src.
Definition: avpacket.c:690
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:99
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4509
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:129
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:4945
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
@ AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT
Definition: log.h:44
AVOptions.
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
Definition: opt.h:279
#define s1
Definition: regdef.h:38
Describe the class of an AVClass context structure.
Definition: log.h:67
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
int frame_size
Audio only.
Definition: codec_par.h:181
int channels
Audio only.
Definition: codec_par.h:166
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:56
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:60
int sample_rate
Audio only.
Definition: codec_par.h:170
List of devices.
Definition: avdevice.h:465
Format I/O context.
Definition: avformat.h:1232
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:645
AVOption.
Definition: opt.h:248
This structure stores compressed data.
Definition: packet.h:346
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:362
Stream structure.
Definition: avformat.h:873
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1038
Definition: alsa.h:48
#define av_log(a,...)
AVPacket * pkt
Definition: movenc.c:59
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:39
double ff_timefilter_update(TimeFilter *self, double system_time, double period)
Update the filter.
Definition: timefilter.c:72
TimeFilter * ff_timefilter_new(double time_base, double period, double bandwidth)
Create a new Delay Locked Loop time filter.
Definition: timefilter.c:46
void ff_timefilter_reset(TimeFilter *self)
Reset the filter.
Definition: timefilter.c:67
enum AVCodecID codec_id
Definition: vaapi_decode.c:369