aboutsummaryrefslogtreecommitdiffstats
path: root/libavdevice/pulse_audio_dec.c
blob: a7b0e5d8c18baa45f4edbd020ffe19b9f945971e (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
/*
 * Pulseaudio input
 * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
 * Copyright 2004-2006 Lennart Poettering
 * Copyright (c) 2014 Michael Niedermayer <michaelni@gmx.at>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include <pulse/rtclock.h>
#include <pulse/error.h>
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "pulse_audio_common.h"
#include "timefilter.h"

#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)

typedef struct PulseData {
    AVClass *class;
    char *server;
    char *name;
    char *stream_name;
    int  sample_rate;
    int  channels;
    int  frame_size;
    int  fragment_size;

    pa_threaded_mainloop *mainloop;
    pa_context *context;
    pa_stream *stream;

    TimeFilter *timefilter;
    int last_period;
} PulseData;


#define CHECK_SUCCESS_GOTO(rerror, expression, label)        \
    do {                                                        \
        if (!(expression)) {                                    \
            rerror = AVERROR_EXTERNAL;                          \
            goto label;                                         \
        }                                                       \
    } while(0);

#define CHECK_DEAD_GOTO(p, rerror, label)                               \
    do {                                                                \
        if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \
            !(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \
            rerror = AVERROR_EXTERNAL;                                  \
            goto label;                                                 \
        }                                                               \
    } while(0);

static void context_state_cb(pa_context *c, void *userdata) {
    PulseData *p = userdata;

    switch (pa_context_get_state(c)) {
        case PA_CONTEXT_READY:
        case PA_CONTEXT_TERMINATED:
        case PA_CONTEXT_FAILED:
            pa_threaded_mainloop_signal(p->mainloop, 0);
            break;
    }
}

static void stream_state_cb(pa_stream *s, void * userdata) {
    PulseData *p = userdata;

    switch (pa_stream_get_state(s)) {
        case PA_STREAM_READY:
        case PA_STREAM_FAILED:
        case PA_STREAM_TERMINATED:
            pa_threaded_mainloop_signal(p->mainloop, 0);
            break;
    }
}

static void stream_request_cb(pa_stream *s, size_t length, void *userdata) {
    PulseData *p = userdata;

    pa_threaded_mainloop_signal(p->mainloop, 0);
}

static void stream_latency_update_cb(pa_stream *s, void *userdata) {
    PulseData *p = userdata;

    pa_threaded_mainloop_signal(p->mainloop, 0);
}

static av_cold int pulse_close(AVFormatContext *s)
{
    PulseData *pd = s->priv_data;

    if (pd->mainloop)
        pa_threaded_mainloop_stop(pd->mainloop);

    if (pd->stream)
        pa_stream_unref(pd->stream);
    pd->stream = NULL;

    if (pd->context) {
        pa_context_disconnect(pd->context);
        pa_context_unref(pd->context);
    }
    pd->context = NULL;

    if (pd->mainloop)
        pa_threaded_mainloop_free(pd->mainloop);
    pd->mainloop = NULL;

    ff_timefilter_destroy(pd->timefilter);
    pd->timefilter = NULL;

    return 0;
}

static av_cold int pulse_read_header(AVFormatContext *s)
{
    PulseData *pd = s->priv_data;
    AVStream *st;
    char *device = NULL;
    int ret;
    enum AVCodecID codec_id =
        s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
    const pa_sample_spec ss = { ff_codec_id_to_pulse_format(codec_id),
                                pd->sample_rate,
                                pd->channels };

    pa_buffer_attr attr = { -1 };

    st = avformat_new_stream(s, NULL);

    if (!st) {
        av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
        return AVERROR(ENOMEM);
    }

    attr.fragsize = pd->fragment_size;

    if (s->filename[0] != '\0' && strcmp(s->filename, "default"))
        device = s->filename;

    if (!(pd->mainloop = pa_threaded_mainloop_new())) {
        pulse_close(s);
        return AVERROR_EXTERNAL;
    }

    if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) {
        pulse_close(s);
        return AVERROR_EXTERNAL;
    }

    pa_context_set_state_callback(pd->context, context_state_cb, pd);

    if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) {
        pulse_close(s);
        return AVERROR(pa_context_errno(pd->context));
    }

    pa_threaded_mainloop_lock(pd->mainloop);

    if (pa_threaded_mainloop_start(pd->mainloop) < 0) {
        ret = -1;
        goto unlock_and_fail;
    }

    for (;;) {
        pa_context_state_t state;

        state = pa_context_get_state(pd->context);

        if (state == PA_CONTEXT_READY)
            break;

        if (!PA_CONTEXT_IS_GOOD(state)) {
            ret = AVERROR(pa_context_errno(pd->context));
            goto unlock_and_fail;
        }

        /* Wait until the context is ready */
        pa_threaded_mainloop_wait(pd->mainloop);
    }

    if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, NULL))) {
        ret = AVERROR(pa_context_errno(pd->context));
        goto unlock_and_fail;
    }

    pa_stream_set_state_callback(pd->stream, stream_state_cb, pd);
    pa_stream_set_read_callback(pd->stream, stream_request_cb, pd);
    pa_stream_set_write_callback(pd->stream, stream_request_cb, pd);
    pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd);

    ret = pa_stream_connect_record(pd->stream, device, &attr,
                                    PA_STREAM_INTERPOLATE_TIMING
                                    |PA_STREAM_ADJUST_LATENCY
                                    |PA_STREAM_AUTO_TIMING_UPDATE);

    if (ret < 0) {
        ret = AVERROR(pa_context_errno(pd->context));
        goto unlock_and_fail;
    }

    for (;;) {
        pa_stream_state_t state;

        state = pa_stream_get_state(pd->stream);

        if (state == PA_STREAM_READY)
            break;

        if (!PA_STREAM_IS_GOOD(state)) {
            ret = AVERROR(pa_context_errno(pd->context));
            goto unlock_and_fail;
        }

        /* Wait until the stream is ready */
        pa_threaded_mainloop_wait(pd->mainloop);
    }

    pa_threaded_mainloop_unlock(pd->mainloop);

    /* take real parameters */
    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
    st->codec->codec_id    = codec_id;
    st->codec->sample_rate = pd->sample_rate;
    st->codec->channels    = pd->channels;
    avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */

    pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate,
                                       1000, 1.5E-6);

    if (!pd->timefilter) {
        pulse_close(s);
        return AVERROR(ENOMEM);
    }

    return 0;

unlock_and_fail:
    pa_threaded_mainloop_unlock(pd->mainloop);

    pulse_close(s);
    return ret;
}

static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
{
    PulseData *pd  = s->priv_data;
    int ret;
    size_t read_length;
    const void *read_data = NULL;
    int64_t dts;
    pa_usec_t latency;
    int negative;

    pa_threaded_mainloop_lock(pd->mainloop);

    CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);

    while (!read_data) {
        int r;

        r = pa_stream_peek(pd->stream, &read_data, &read_length);
        CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);

        if (read_length <= 0) {
            pa_threaded_mainloop_wait(pd->mainloop);
            CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
        } else if (!read_data) {
            /* There's a hole in the stream, skip it. We could generate
                * silence, but that wouldn't work for compressed streams. */
            r = pa_stream_drop(pd->stream);
            CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
        }
    }

    if (av_new_packet(pkt, read_length) < 0) {
        ret = AVERROR(ENOMEM);
        goto unlock_and_fail;
    }

    dts = av_gettime();
    pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));

    if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) {
        enum AVCodecID codec_id =
            s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
        int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels);
        int frame_duration = read_length / frame_size;


        if (negative) {
            dts += latency;
        } else
            dts -= latency;
        pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period);

        pd->last_period = frame_duration;
    } else {
        av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n");
    }

    memcpy(pkt->data, read_data, read_length);
    pa_stream_drop(pd->stream);

    pa_threaded_mainloop_unlock(pd->mainloop);
    return 0;

unlock_and_fail:
    pa_threaded_mainloop_unlock(pd->mainloop);
    return ret;
}

static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
{
    PulseData *s = h->priv_data;
    return ff_pulse_audio_get_devices(device_list, s->server, 0);
}

#define OFFSET(a) offsetof(PulseData, a)
#define D AV_OPT_FLAG_DECODING_PARAM

static const AVOption options[] = {
    { "server",        "set PulseAudio server",                             OFFSET(server),        AV_OPT_TYPE_STRING, {.str = NULL},     0, 0, D },
    { "name",          "set application name",                              OFFSET(name),          AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT},  0, 0, D },
    { "stream_name",   "set stream description",                            OFFSET(stream_name),   AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
    { "sample_rate",   "set sample rate in Hz",                             OFFSET(sample_rate),   AV_OPT_TYPE_INT,    {.i64 = 48000},    1, INT_MAX, D },
    { "channels",      "set number of audio channels",                      OFFSET(channels),      AV_OPT_TYPE_INT,    {.i64 = 2},        1, INT_MAX, D },
    { "frame_size",    "set number of bytes per frame",                     OFFSET(frame_size),    AV_OPT_TYPE_INT,    {.i64 = 1024},     1, INT_MAX, D },
    { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT,    {.i64 = -1},      -1, INT_MAX, D },
    { NULL },
};

static const AVClass pulse_demuxer_class = {
    .class_name     = "Pulse demuxer",
    .item_name      = av_default_item_name,
    .option         = options,
    .version        = LIBAVUTIL_VERSION_INT,
    .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
};

AVInputFormat ff_pulse_demuxer = {
    .name           = "pulse",
    .long_name      = NULL_IF_CONFIG_SMALL("Pulse audio input"),
    .priv_data_size = sizeof(PulseData),
    .read_header    = pulse_read_header,
    .read_packet    = pulse_read_packet,
    .read_close     = pulse_close,
    .get_device_list = pulse_get_device_list,
    .flags          = AVFMT_NOFILE,
    .priv_class     = &pulse_demuxer_class,
};