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: 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 */
39
40 #include <alsa/asoundlib.h>
41
44
45
49
51 {
56 int res;
57
61 }
63
66 res =
ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
70 "sample rate %d not available, nearest is %d\n",
73 }
75
76 return res;
77
81 }
82
84 {
86 int res;
89
94
101 }
102 while ((res = snd_pcm_writei(s->
h, buf, size)) < 0) {
103 if (res == -EAGAIN) {
104
106 }
107
110 snd_strerror(res));
111
113 }
114 }
115
116 return 0;
117 }
118
121 {
124
125 /* ff_alsa_open() should have accepted only supported formats */
129 /* set only used fields */
130 pkt.
data = (*frame)->data[0];
132 pkt.
dts = (*frame)->pkt_dts;
133 pkt.
duration = (*frame)->pkt_duration;
135 }
136
137 static void
139 int64_t *dts, int64_t *wall)
140 {
142 snd_pcm_sframes_t delay = 0;
144 snd_pcm_delay(s->
h, &delay);
146 }
147
149 {
151 }
152
158 };
159
173 .priv_class = &alsa_muxer_class,
174 };
static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int unqueue)
This structure describes decoded (raw) audio or video data.
#define LIBAVUTIL_VERSION_INT
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
const char * av_default_item_name(void *ptr)
Return the context name.
int64_t timestamp
current timestamp, without latency applied.
AVOutputFormat ff_alsa_muxer
ALSA input and output: definitions and structures.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
void(* reorder_func)(const void *, void *, int)
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
AVStream ** streams
A list of all streams in the file.
static av_cold int audio_write_header(AVFormatContext *s1)
static int audio_write_frame(AVFormatContext *s1, int stream_index, AVFrame **frame, unsigned flags)
int ff_alsa_extend_reorder_buf(AlsaData *s, int min_size)
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Main libavdevice API header.
AVCodecID
Identify the syntax and semantics of the bitstream.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
enum AVMediaType codec_type
General type of the encoded data.
common internal API header
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
static int write_trailer(AVFormatContext *s1)
static const AVClass alsa_muxer_class
int64_t av_gettime(void)
Get the current time in microseconds.
int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
Try to recover from ALSA buffer underrun.
av_cold int ff_alsa_close(AVFormatContext *s1)
Close the ALSA PCM.
static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
static void audio_get_output_timestamp(AVFormatContext *s1, int stream, int64_t *dts, int64_t *wall)
Describe the class of an AVClass context structure.
int ff_alsa_get_device_list(AVDeviceInfoList *device_list, snd_pcm_stream_t stream_type)
int sample_rate
Audio only.
void * priv_data
Format private data.
static void write_header(FFV1Context *f)
int64_t dts
Decompression timestamp in AVStream->time_base units; the time at which the packet is decompressed...
int frame_size
bytes per sample * channels
AVCodecParameters * codecpar
Codec parameters associated with this stream.
This structure stores compressed data.
int reorder_buf_size
in frames
#define AV_NOPTS_VALUE
Undefined timestamp value.
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.