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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
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  */
23 /**
24  * @file
25  * ALSA input and output: output
26  * @author Luca Abeni ( lucabe72 email it )
27  * @author Benoit Fouet ( benoit fouet free fr )
28  *
29  * This avdevice encoder can play audio to an ALSA (Advanced Linux
30  * Sound Architecture) device.
31  *
32  * The filename parameter is the name of an ALSA PCM device capable of
33  * capture, for example "default" or "plughw:1"; see the ALSA documentation
34  * for naming conventions. The empty string is equivalent to "default".
35  *
36  * The playback period is set to the lower value available for the device,
37  * which gives a low latency suitable for real-time playback.
38  */
40 #include <alsa/asoundlib.h>
42 #include "libavutil/internal.h"
43 #include "libavutil/time.h"
46 #include "libavformat/internal.h"
47 #include "avdevice.h"
48 #include "alsa.h"
51 {
52  AlsaData *s = s1->priv_data;
53  AVStream *st = NULL;
54  unsigned int sample_rate;
55  enum AVCodecID codec_id;
56  int res;
58  if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
59  av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
60  return AVERROR(EINVAL);
61  }
62  st = s1->streams[0];
64  sample_rate = st->codecpar->sample_rate;
65  codec_id = st->codecpar->codec_id;
66  res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
67  st->codecpar->channels, &codec_id);
68  if (sample_rate != st->codecpar->sample_rate) {
69  av_log(s1, AV_LOG_ERROR,
70  "sample rate %d not available, nearest is %d\n",
71  st->codecpar->sample_rate, sample_rate);
72  goto fail;
73  }
74  avpriv_set_pts_info(st, 64, 1, sample_rate);
76  return res;
78 fail:
79  snd_pcm_close(s->h);
80  return AVERROR(EIO);
81 }
84 {
85  AlsaData *s = s1->priv_data;
86  int res;
87  int size = pkt->size;
88  uint8_t *buf = pkt->data;
90  size /= s->frame_size;
91  if (pkt->dts != AV_NOPTS_VALUE)
92  s->timestamp = pkt->dts;
93  s->timestamp += pkt->duration ? pkt->duration : size;
95  if (s->reorder_func) {
96  if (size > s->reorder_buf_size)
97  if (ff_alsa_extend_reorder_buf(s, size))
98  return AVERROR(ENOMEM);
99  s->reorder_func(buf, s->reorder_buf, size);
100  buf = s->reorder_buf;
101  }
102  while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
103  if (res == -EAGAIN) {
105  return AVERROR(EAGAIN);
106  }
108  if (ff_alsa_xrun_recover(s1, res) < 0) {
109  av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
110  snd_strerror(res));
112  return AVERROR(EIO);
113  }
114  }
116  return 0;
117 }
119 static int audio_write_frame(AVFormatContext *s1, int stream_index,
120  AVFrame **frame, unsigned flags)
121 {
122  AlsaData *s = s1->priv_data;
123  AVPacket pkt;
125  /* ff_alsa_open() should have accepted only supported formats */
126  if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
127  return av_sample_fmt_is_planar(s1->streams[stream_index]->codecpar->format) ?
129  /* set only used fields */
130  pkt.data = (*frame)->data[0];
131  pkt.size = (*frame)->nb_samples * s->frame_size;
132  pkt.dts = (*frame)->pkt_dts;
133  pkt.duration = (*frame)->pkt_duration;
134  return audio_write_packet(s1, &pkt);
135 }
137 static void
139  int64_t *dts, int64_t *wall)
140 {
141  AlsaData *s = s1->priv_data;
142  snd_pcm_sframes_t delay = 0;
143  *wall = av_gettime();
144  snd_pcm_delay(s->h, &delay);
145  *dts = s->timestamp - delay;
146 }
149 {
150  return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
151 }
153 static const AVClass alsa_muxer_class = {
154  .class_name = "ALSA outdev",
155  .item_name = av_default_item_name,
156  .version = LIBAVUTIL_VERSION_INT,
158 };
161  .name = "alsa",
162  .long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
163  .priv_data_size = sizeof(AlsaData),
164  .audio_codec = DEFAULT_CODEC_ID,
165  .video_codec = AV_CODEC_ID_NONE,
169  .write_uncoded_frame = audio_write_frame,
170  .get_device_list = audio_get_device_list,
171  .get_output_timestamp = audio_get_output_timestamp,
172  .flags = AVFMT_NOFILE,
173  .priv_class = &alsa_muxer_class,
174 };
static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int unqueue)
Definition: ffmpeg.c:726
#define NULL
Definition: coverity.c:32
This structure describes decoded (raw) audio or video data.
Definition: frame.h:314
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:4893
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:60
int size
Definition: packet.h:364
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
int64_t timestamp
current timestamp, without latency applied.
Definition: alsa.h:60
static AVPacket pkt
AVOutputFormat ff_alsa_muxer
Definition: alsa_enc.c:160
ALSA input and output: definitions and structures.
Format I/O context.
Definition: avformat.h:1239
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:88
Query whether the feature is possible on this stream.
Definition: internal.h:772
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:381
AVStream ** streams
A list of all streams in the file.
Definition: avformat.h:1307
static av_cold int audio_write_header(AVFormatContext *s1)
Definition: alsa_enc.c:50
uint8_t * data
Definition: packet.h:363
static int audio_write_frame(AVFormatContext *s1, int stream_index, AVFrame **frame, unsigned flags)
Definition: alsa_enc.c:119
ptrdiff_t size
Definition: opengl_enc.c:100
int ff_alsa_extend_reorder_buf(AlsaData *s, int min_size)
Definition: alsa.c:335
#define av_log(a,...)
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
Main libavdevice API header.
Identify the syntax and semantics of the bitstream.
Definition: codec_id.h:46
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
Definition: alsa.h:42
Definition: alsa.h:48
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:115
static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
Definition: alsa_enc.c:83
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:56
#define fail()
Definition: checkasm.h:123
common internal API header
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1295
void(* reorder_func)(const void *, void *, int)
Definition: alsa.h:57
static int write_trailer(AVFormatContext *s1)
Definition: v4l2enc.c:98
const char * name
Definition: avformat.h:500
static const AVClass alsa_muxer_class
Definition: alsa_enc.c:153
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
#define s(width, name)
Definition: cbs_vp9.c:257
enum AVCodecID codec_id
Definition: vaapi_decode.c:369
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:39
Stream structure.
Definition: avformat.h:880
void * reorder_buf
Definition: alsa.h:58
int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
Try to recover from ALSA buffer underrun.
Definition: alsa.c:314
av_cold int ff_alsa_close(AVFormatContext *s1)
Close the ALSA PCM.
Definition: alsa.c:299
static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
Definition: alsa_enc.c:148
static void audio_get_output_timestamp(AVFormatContext *s1, int stream, int64_t *dts, int64_t *wall)
Definition: alsa_enc.c:138
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 s1
Definition: regdef.h:38
List of devices.
Definition: avdevice.h:460
#define flags(name, subs,...)
Definition: cbs_av1.c:561
int sample_rate
Audio only.
Definition: codec_par.h:170
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:458
void * priv_data
Format private data.
Definition: avformat.h:1267
static void write_header(FFV1Context *f)
Definition: ffv1enc.c:346
snd_pcm_t * h
Definition: alsa.h:50
int channels
Audio only.
Definition: codec_par.h:166
int64_t dts
Decompression timestamp in AVStream->time_base units; the time at which the packet is decompressed...
Definition: packet.h:362
int frame_size
bytes per sample * channels
Definition: alsa.h:51
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1045
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
This structure stores compressed data.
Definition: packet.h:340
int reorder_buf_size
in frames
Definition: alsa.h:59
Undefined timestamp value.
Definition: avutil.h:248
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