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 
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 };
AVERROR
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
opt.h
avformat_new_stream
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4376
AVCodecParameters::codec_type
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:56
AVOption
AVOption.
Definition: opt.h:248
mathematics.h
sample_rate
sample_rate
Definition: ffmpeg_filter.c:156
ff_alsa_demuxer
const AVInputFormat ff_alsa_demuxer
Definition: alsa_dec.c:160
AVCodecParameters::channels
int channels
Audio only.
Definition: codec_par.h:166
fail
#define fail()
Definition: checkasm.h:134
read_close
static av_cold int read_close(AVFormatContext *ctx)
Definition: libcdio.c:142
ff_timefilter_new
TimeFilter * ff_timefilter_new(double time_base, double period, double bandwidth)
Create a new Delay Locked Loop time filter.
Definition: timefilter.c:46
AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT
@ AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT
Definition: log.h:44
pkt
AVPacket * pkt
Definition: movenc.c:59
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:181
AVInputFormat
Definition: avformat.h:621
av_cold
#define av_cold
Definition: attributes.h:90
AVCodecParameters::frame_size
int frame_size
Audio only.
Definition: codec_par.h:181
s
#define s(width, name)
Definition: cbs_vp9.c:257
av_new_packet
int av_new_packet(AVPacket *pkt, int size)
Allocate the payload of a packet and initialize its fields with default values.
Definition: avpacket.c:94
AVInputFormat::name
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:626
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
s1
#define s1
Definition: regdef.h:38
channels
channels
Definition: aptx.h:33
ff_alsa_open
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
codec_id
enum AVCodecID codec_id
Definition: vaapi_decode.c:369
AVFormatContext
Format I/O context.
Definition: avformat.h:1097
internal.h
AVStream::codecpar
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:984
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
read_header
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:527
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
NULL
#define NULL
Definition: coverity.c:32
alsa_demuxer_class
static const AVClass alsa_demuxer_class
Definition: alsa_dec.c:152
AlsaData
Definition: alsa.h:48
audio_read_header
static av_cold int audio_read_header(AVFormatContext *s1)
Definition: alsa_dec.c:60
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
time.h
av_packet_move_ref
void av_packet_move_ref(AVPacket *dst, AVPacket *src)
Move every field in src to dst and reset src.
Definition: avpacket.c:459
ff_alsa_close
av_cold int ff_alsa_close(AVFormatContext *s1)
Close the ALSA PCM.
Definition: alsa.c:303
AVCodecParameters::sample_rate
int sample_rate
Audio only.
Definition: codec_par.h:170
AVCodecID
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: codec_id.h:47
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
avpriv_set_pts_info
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:4801
AVFMT_NOFILE
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:463
avdevice.h
ff_timefilter_reset
void ff_timefilter_reset(TimeFilter *self)
Reset the filter.
Definition: timefilter.c:67
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:358
internal.h
AV_OPT_FLAG_DECODING_PARAM
#define AV_OPT_FLAG_DECODING_PARAM
a generic parameter which can be set by the user for demuxing or decoding
Definition: opt.h:279
ff_timefilter_update
double ff_timefilter_update(TimeFilter *self, double system_time, double period)
Update the filter.
Definition: timefilter.c:72
av_rescale
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
alsa.h
ret
ret
Definition: filter_design.txt:187
read_packet
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
AVStream
Stream structure.
Definition: avformat.h:837
AVClass::class_name
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
AVDeviceInfoList
List of devices.
Definition: avdevice.h:465
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:225
audio_get_device_list
static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
Definition: alsa_dec.c:141
ff_alsa_xrun_recover
int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
Try to recover from ALSA buffer underrun.
Definition: alsa.c:319
av_gettime
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:39
options
static const AVOption options[]
Definition: alsa_dec.c:146
AVCodecParameters::codec_id
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:60
AVPacket
This structure stores compressed data.
Definition: packet.h:342
ff_alsa_get_device_list
int ff_alsa_get_device_list(AVDeviceInfoList *device_list, snd_pcm_stream_t stream_type)
Definition: alsa.c:357
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:561
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
h
h
Definition: vp9dsp_template.c:2038
audio_read_packet
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: alsa_dec.c:101