FFmpeg  2.1.1
alsa-audio-dec.c
Go to the documentation of this file.
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:279
AVOption.
Definition: opt.h:253
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:478
#define LIBAVUTIL_VERSION_INT
Definition: avcodec.h:820
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:3922
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:686
int size
Definition: avcodec.h:1064
void av_log(void *avcl, int level, const char *fmt,...) av_printf_format(3
Send the specified message to the log if the level is less than or equal to the current av_log_level...
#define av_cold
Definition: avcodec.h:653
Format I/O context.
Definition: avformat.h:968
static uint8_t * res
Definition: ffhash.c:43
const char * av_default_item_name(void *ctx)
Return the context name.
Definition: log.c:145
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
av_cold int ff_alsa_close(AVFormatContext *s1)
Close the ALSA PCM.
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:145
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:347
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:3348
const OptionDef options[]
Definition: ffserver.c:4682
static av_cold int audio_read_header(AVFormatContext *s1)
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:83
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: avcodec.h:102
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: avcodec.h:4147
void * priv_data
Format private data.
Definition: avformat.h:988
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:151
enum AVCodecID codec_id
Definition: mov_chan.c:433
goto fail
Definition: avfilter.c:963
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:39
int64_t av_rescale(int64_t a, int64_t b, int64_t c) av_const
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:1088
ret
Definition: avfilter.c:961
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.
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:592
AVInputFormat ff_alsa_demuxer
Stream structure.
Definition: avformat.h:667
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
Definition: avcodec.h:1154
enum AVCodecID codec_id
Definition: avcodec.h:1157
int sample_rate
samples per second
Definition: avcodec.h:1873
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
Main libavdevice API header.
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
uint8_t * data
Definition: avcodec.h:1063
int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
Try to recover from ALSA buffer underrun.
#define s1
Definition: regdef.h:38
int period_size
preferred size for reads and writes, in frames
Definition: alsa-audio.h:52
static int flags
Definition: cpu.c:45
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
Definition: opt.h:284
double ff_timefilter_update(TimeFilter *self, double system_time, double period)
Update the filter.
Definition: timefilter.c:72
int last_period
Definition: alsa-audio.h:55
int channels
number of audio channels
Definition: avcodec.h:1874
#define AVERROR(e)
snd_pcm_t * h
Definition: alsa-audio.h:50
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.
static AVPacket pkt
Definition: demuxing.c:52
This structure stores compressed data.
Definition: avcodec.h:1040
int64_t pts
Presentation timestamp in AVStream-&gt;time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1056