FFmpeg
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 
67  st = avformat_new_stream(s1, NULL);
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;
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 (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
109  return AVERROR(EIO);
110  }
111 
112  while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
113  if (res == -EAGAIN) {
114  av_packet_unref(pkt);
115 
116  return AVERROR(EAGAIN);
117  }
118  if (ff_alsa_xrun_recover(s1, res) < 0) {
119  av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
120  snd_strerror(res));
121  av_packet_unref(pkt);
122 
123  return AVERROR(EIO);
124  }
126  }
127 
128  dts = av_gettime();
129  snd_pcm_delay(s->h, &delay);
130  dts -= av_rescale(delay + res, 1000000, s->sample_rate);
131  pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
132  s->last_period = res;
133 
134  pkt->size = res * s->frame_size;
135 
136  return 0;
137 }
138 
140 {
141  return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
142 }
143 
144 static const AVOption options[] = {
145  { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
146  { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
147  { NULL },
148 };
149 
150 static const AVClass alsa_demuxer_class = {
151  .class_name = "ALSA demuxer",
152  .item_name = av_default_item_name,
153  .option = options,
154  .version = LIBAVUTIL_VERSION_INT,
156 };
157 
159  .name = "alsa",
160  .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
161  .priv_data_size = sizeof(AlsaData),
165  .get_device_list = audio_get_device_list,
166  .flags = AVFMT_NOFILE,
167  .priv_class = &alsa_demuxer_class,
168 };
#define NULL
Definition: coverity.c:32
AVOption.
Definition: opt.h:246
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
void ff_timefilter_reset(TimeFilter *self)
Reset the filter.
Definition: timefilter.c:67
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:4882
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: alsa_dec.c:101
channels
Definition: aptx.c:30
enum AVCodecID codec_id
Definition: qsv.c:72
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: avcodec.h:3953
int size
Definition: avcodec.h:1478
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
static AVPacket pkt
int frame_size
Audio only.
Definition: avcodec.h:4074
ALSA input and output: definitions and structures.
Format I/O context.
Definition: avformat.h:1358
static const AVOption options[]
Definition: alsa_dec.c:144
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
#define av_cold
Definition: attributes.h:82
AVOptions.
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4455
uint8_t * data
Definition: avcodec.h:1477
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
#define av_log(a,...)
Main libavdevice API header.
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:86
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: avcodec.h:215
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
Definition: alsa.h:48
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
enum AVMediaType codec_type
General type of the encoded data.
Definition: avcodec.h:3949
#define fail()
Definition: checkasm.h:120
common internal API header
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
enum AVCodecID audio_codec_id
Forced audio codec_id.
Definition: avformat.h:1550
static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
Definition: alsa_dec.c:139
int channels
number of channels set by user
Definition: alsa.h:54
#define s(width, name)
Definition: cbs_vp9.c:257
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:530
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:39
Stream structure.
Definition: avformat.h:881
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
sample_rate
TimeFilter * ff_timefilter_new(double time_base, double period, double bandwidth)
Create a new Delay Locked Loop time filter.
Definition: timefilter.c:46
int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
Try to recover from ALSA buffer underrun.
Definition: alsa.c:314
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:599
av_cold int ff_alsa_close(AVFormatContext *s1)
Close the ALSA PCM.
Definition: alsa.c:299
TimeFilter * timefilter
Definition: alsa.h:56
Describe the class of an AVClass context structure.
Definition: log.h:67
int ff_alsa_get_device_list(AVDeviceInfoList *device_list, snd_pcm_stream_t stream_type)
Definition: alsa.c:352
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
Definition: opt.h:277
#define s1
Definition: regdef.h:38
List of devices.
Definition: avdevice.h:460
#define flags(name, subs,...)
Definition: cbs_av1.c:561
int period_size
preferred size for reads and writes, in frames
Definition: alsa.h:52
int sample_rate
Audio only.
Definition: avcodec.h:4063
double ff_timefilter_update(TimeFilter *self, double system_time, double period)
Update the filter.
Definition: timefilter.c:72
int last_period
Definition: alsa.h:55
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:463
void * priv_data
Format private data.
Definition: avformat.h:1386
static av_cold int audio_read_header(AVFormatContext *s1)
Definition: alsa_dec.c:60
snd_pcm_t * h
Definition: alsa.h:50
int channels
Audio only.
Definition: avcodec.h:4059
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:654
int frame_size
bytes per sample * channels
Definition: alsa.h:51
int sample_rate
sample rate set by user
Definition: alsa.h:53
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1028
static const AVClass alsa_demuxer_class
Definition: alsa_dec.c:150
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
AVInputFormat ff_alsa_demuxer
Definition: alsa_dec.c:158
This structure stores compressed data.
Definition: avcodec.h:1454
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1470
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