alsa-audio-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 allows to 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 #include "libavformat/internal.h"
50 #include "libavutil/opt.h"
51 #include "libavutil/mathematics.h"
52 #include "libavutil/time.h"
53 
54 #include "avdevice.h"
55 #include "alsa-audio.h"
56 
58 {
59  AlsaData *s = s1->priv_data;
60  AVStream *st;
61  int ret;
62  enum AVCodecID codec_id;
63 
64  st = avformat_new_stream(s1, NULL);
65  if (!st) {
66  av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
67 
68  return AVERROR(ENOMEM);
69  }
70  codec_id = s1->audio_codec_id;
71 
72  ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
73  &codec_id);
74  if (ret < 0) {
75  return AVERROR(EIO);
76  }
77 
78  /* take real parameters */
80  st->codec->codec_id = codec_id;
81  st->codec->sample_rate = s->sample_rate;
82  st->codec->channels = s->channels;
83  avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
84  /* microseconds instead of seconds, MHz instead of Hz */
85  s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
86  s->period_size, 1.5E-6);
87  if (!s->timefilter)
88  goto fail;
89 
90  return 0;
91 
92 fail:
93  snd_pcm_close(s->h);
94  return AVERROR(EIO);
95 }
96 
98 {
99  AlsaData *s = s1->priv_data;
100  int res;
101  int64_t dts;
102  snd_pcm_sframes_t delay = 0;
103 
104  if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
105  return AVERROR(EIO);
106  }
107 
108  while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
109  if (res == -EAGAIN) {
110  av_free_packet(pkt);
111 
112  return AVERROR(EAGAIN);
113  }
114  if (ff_alsa_xrun_recover(s1, res) < 0) {
115  av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
116  snd_strerror(res));
117  av_free_packet(pkt);
118 
119  return AVERROR(EIO);
120  }
122  }
123 
124  dts = av_gettime();
125  snd_pcm_delay(s->h, &delay);
126  dts -= av_rescale(delay + res, 1000000, s->sample_rate);
127  pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
128  s->last_period = res;
129 
130  pkt->size = res * s->frame_size;
131 
132  return 0;
133 }
134 
135 static const AVOption options[] = {
136  { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
137  { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
138  { NULL },
139 };
140 
141 static const AVClass alsa_demuxer_class = {
142  .class_name = "ALSA demuxer",
143  .item_name = av_default_item_name,
144  .option = options,
145  .version = LIBAVUTIL_VERSION_INT,
146 };
147 
149  .name = "alsa",
150  .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
151  .priv_data_size = sizeof(AlsaData),
155  .flags = AVFMT_NOFILE,
156  .priv_class = &alsa_demuxer_class,
157 };
const char * s
Definition: avisynth_c.h:668
void av_free_packet(AVPacket *pkt)
Free a packet.
Definition: avpacket.c:242
AVOption.
Definition: opt.h:251
av_default_item_name
void ff_timefilter_reset(TimeFilter *self)
Reset the filter.
Definition: timefilter.c:63
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.
Format I/O context.
Definition: avformat.h:944
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:55
#define av_cold
Definition: attributes.h:78
AVOptions.
static AVPacket pkt
Definition: demuxing.c:56
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
av_cold int ff_alsa_close(AVFormatContext *s1)
Close the ALSA PCM.
uint8_t * data
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
static av_cold int audio_read_header(AVFormatContext *s1)
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:73
AVCodecID
Identify the syntax and semantics of the bitstream.
static const AVOption options[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:246
enum AVCodecID codec_id
Definition: mov_chan.c:433
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:662
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:118
enum AVCodecID audio_codec_id
Forced audio codec_id.
Definition: avformat.h:1063
ret
Definition: avfilter.c:821
int channels
number of channels set by user
Definition: alsa-audio.h:54
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.
LIBAVUTIL_VERSION_INT
Definition: eval.c:55
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:517
AVInputFormat ff_alsa_demuxer
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:39
Stream structure.
Definition: avformat.h:643
NULL
Definition: eval.c:55
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
enum AVMediaType codec_type
enum AVCodecID codec_id
int sample_rate
samples per second
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:148
TimeFilter * timefilter
Definition: alsa-audio.h:56
static int read_packet(AVFormatContext *ctx, AVPacket *pkt)
Definition: libcdio.c:114
static const AVClass alsa_demuxer_class
Describe the class of an AVClass context structure.
Definition: log.h:50
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
Definition: opt.h:282
int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
Try to recover from ALSA buffer underrun.
#define s1
Definition: regdef.h:38
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer 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.Buffer references ownership and permissions
struct AlsaData AlsaData
int period_size
preferred size for reads and writes, in frames
Definition: alsa-audio.h:52
static int flags
Definition: cpu.c:23
double ff_timefilter_update(TimeFilter *self, double system_time, double period)
Update the filter.
Definition: timefilter.c:68
int last_period
Definition: alsa-audio.h:55
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:345
int channels
number of audio channels
void * priv_data
Format private data.
Definition: avformat.h:964
snd_pcm_t * h
Definition: alsa-audio.h:50
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:461
int frame_size
bytes per sample * channels
Definition: alsa-audio.h:51
int sample_rate
sample rate set by user
Definition: alsa-audio.h:53
ALSA input and output: definitions and structures.
This structure stores compressed data.
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...