1/*
2 * Pulseaudio input
3 * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
4 * Copyright 2004-2006 Lennart Poettering
5 * Copyright (c) 2014 Michael Niedermayer <michaelni@gmx.at>
6 *
7 * This file is part of FFmpeg.
8 *
9 * FFmpeg is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Lesser General Public
11 * License as published by the Free Software Foundation; either
12 * version 2.1 of the License, or (at your option) any later version.
13 *
14 * FFmpeg is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
17 * Lesser General Public License for more details.
18 *
19 * You should have received a copy of the GNU Lesser General Public
20 * License along with FFmpeg; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 */
23
24#include <pulse/rtclock.h>
25#include <pulse/error.h>
26#include "libavformat/avformat.h"
27#include "libavformat/internal.h"
28#include "libavutil/opt.h"
29#include "libavutil/time.h"
30#include "pulse_audio_common.h"
31#include "timefilter.h"
32
33#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
34
35typedef struct PulseData {
36    AVClass *class;
37    char *server;
38    char *name;
39    char *stream_name;
40    int  sample_rate;
41    int  channels;
42    int  frame_size;
43    int  fragment_size;
44
45    pa_threaded_mainloop *mainloop;
46    pa_context *context;
47    pa_stream *stream;
48
49    TimeFilter *timefilter;
50    int last_period;
51} PulseData;
52
53
54#define CHECK_SUCCESS_GOTO(rerror, expression, label)        \
55    do {                                                        \
56        if (!(expression)) {                                    \
57            rerror = AVERROR_EXTERNAL;                          \
58            goto label;                                         \
59        }                                                       \
60    } while(0);
61
62#define CHECK_DEAD_GOTO(p, rerror, label)                               \
63    do {                                                                \
64        if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \
65            !(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \
66            rerror = AVERROR_EXTERNAL;                                  \
67            goto label;                                                 \
68        }                                                               \
69    } while(0);
70
71static void context_state_cb(pa_context *c, void *userdata) {
72    PulseData *p = userdata;
73
74    switch (pa_context_get_state(c)) {
75        case PA_CONTEXT_READY:
76        case PA_CONTEXT_TERMINATED:
77        case PA_CONTEXT_FAILED:
78            pa_threaded_mainloop_signal(p->mainloop, 0);
79            break;
80    }
81}
82
83static void stream_state_cb(pa_stream *s, void * userdata) {
84    PulseData *p = userdata;
85
86    switch (pa_stream_get_state(s)) {
87        case PA_STREAM_READY:
88        case PA_STREAM_FAILED:
89        case PA_STREAM_TERMINATED:
90            pa_threaded_mainloop_signal(p->mainloop, 0);
91            break;
92    }
93}
94
95static void stream_request_cb(pa_stream *s, size_t length, void *userdata) {
96    PulseData *p = userdata;
97
98    pa_threaded_mainloop_signal(p->mainloop, 0);
99}
100
101static void stream_latency_update_cb(pa_stream *s, void *userdata) {
102    PulseData *p = userdata;
103
104    pa_threaded_mainloop_signal(p->mainloop, 0);
105}
106
107static av_cold int pulse_close(AVFormatContext *s)
108{
109    PulseData *pd = s->priv_data;
110
111    if (pd->mainloop)
112        pa_threaded_mainloop_stop(pd->mainloop);
113
114    if (pd->stream)
115        pa_stream_unref(pd->stream);
116    pd->stream = NULL;
117
118    if (pd->context) {
119        pa_context_disconnect(pd->context);
120        pa_context_unref(pd->context);
121    }
122    pd->context = NULL;
123
124    if (pd->mainloop)
125        pa_threaded_mainloop_free(pd->mainloop);
126    pd->mainloop = NULL;
127
128    ff_timefilter_destroy(pd->timefilter);
129    pd->timefilter = NULL;
130
131    return 0;
132}
133
134static av_cold int pulse_read_header(AVFormatContext *s)
135{
136    PulseData *pd = s->priv_data;
137    AVStream *st;
138    char *device = NULL;
139    int ret;
140    enum AVCodecID codec_id =
141        s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
142    const pa_sample_spec ss = { ff_codec_id_to_pulse_format(codec_id),
143                                pd->sample_rate,
144                                pd->channels };
145
146    pa_buffer_attr attr = { -1 };
147
148    st = avformat_new_stream(s, NULL);
149
150    if (!st) {
151        av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
152        return AVERROR(ENOMEM);
153    }
154
155    attr.fragsize = pd->fragment_size;
156
157    if (strcmp(s->filename, "default"))
158        device = s->filename;
159
160    if (!(pd->mainloop = pa_threaded_mainloop_new())) {
161        pulse_close(s);
162        return AVERROR_EXTERNAL;
163    }
164
165    if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) {
166        pulse_close(s);
167        return AVERROR_EXTERNAL;
168    }
169
170    pa_context_set_state_callback(pd->context, context_state_cb, pd);
171
172    if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) {
173        pulse_close(s);
174        return AVERROR(pa_context_errno(pd->context));
175    }
176
177    pa_threaded_mainloop_lock(pd->mainloop);
178
179    if (pa_threaded_mainloop_start(pd->mainloop) < 0) {
180        ret = -1;
181        goto unlock_and_fail;
182    }
183
184    for (;;) {
185        pa_context_state_t state;
186
187        state = pa_context_get_state(pd->context);
188
189        if (state == PA_CONTEXT_READY)
190            break;
191
192        if (!PA_CONTEXT_IS_GOOD(state)) {
193            ret = AVERROR(pa_context_errno(pd->context));
194            goto unlock_and_fail;
195        }
196
197        /* Wait until the context is ready */
198        pa_threaded_mainloop_wait(pd->mainloop);
199    }
200
201    if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, NULL))) {
202        ret = AVERROR(pa_context_errno(pd->context));
203        goto unlock_and_fail;
204    }
205
206    pa_stream_set_state_callback(pd->stream, stream_state_cb, pd);
207    pa_stream_set_read_callback(pd->stream, stream_request_cb, pd);
208    pa_stream_set_write_callback(pd->stream, stream_request_cb, pd);
209    pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd);
210
211    ret = pa_stream_connect_record(pd->stream, device, &attr,
212                                    PA_STREAM_INTERPOLATE_TIMING
213                                    |PA_STREAM_ADJUST_LATENCY
214                                    |PA_STREAM_AUTO_TIMING_UPDATE);
215
216    if (ret < 0) {
217        ret = AVERROR(pa_context_errno(pd->context));
218        goto unlock_and_fail;
219    }
220
221    for (;;) {
222        pa_stream_state_t state;
223
224        state = pa_stream_get_state(pd->stream);
225
226        if (state == PA_STREAM_READY)
227            break;
228
229        if (!PA_STREAM_IS_GOOD(state)) {
230            ret = AVERROR(pa_context_errno(pd->context));
231            goto unlock_and_fail;
232        }
233
234        /* Wait until the stream is ready */
235        pa_threaded_mainloop_wait(pd->mainloop);
236    }
237
238    pa_threaded_mainloop_unlock(pd->mainloop);
239
240    /* take real parameters */
241    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
242    st->codec->codec_id    = codec_id;
243    st->codec->sample_rate = pd->sample_rate;
244    st->codec->channels    = pd->channels;
245    avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
246
247    pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate,
248                                       1000, 1.5E-6);
249
250    if (!pd->timefilter) {
251        pulse_close(s);
252        return AVERROR(ENOMEM);
253    }
254
255    return 0;
256
257unlock_and_fail:
258    pa_threaded_mainloop_unlock(pd->mainloop);
259
260    pulse_close(s);
261    return ret;
262}
263
264static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
265{
266    PulseData *pd  = s->priv_data;
267    int ret;
268    size_t read_length;
269    const void *read_data = NULL;
270    int64_t dts;
271    pa_usec_t latency;
272    int negative;
273
274    pa_threaded_mainloop_lock(pd->mainloop);
275
276    CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
277
278    while (!read_data) {
279        int r;
280
281        r = pa_stream_peek(pd->stream, &read_data, &read_length);
282        CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
283
284        if (read_length <= 0) {
285            pa_threaded_mainloop_wait(pd->mainloop);
286            CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
287        } else if (!read_data) {
288            /* There's a hole in the stream, skip it. We could generate
289                * silence, but that wouldn't work for compressed streams. */
290            r = pa_stream_drop(pd->stream);
291            CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
292        }
293    }
294
295    if (av_new_packet(pkt, read_length) < 0) {
296        ret = AVERROR(ENOMEM);
297        goto unlock_and_fail;
298    }
299
300    dts = av_gettime();
301    pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));
302
303    if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) {
304        enum AVCodecID codec_id =
305            s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
306        int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels);
307        int frame_duration = read_length / frame_size;
308
309
310        if (negative) {
311            dts += latency;
312        } else
313            dts -= latency;
314        pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period);
315
316        pd->last_period = frame_duration;
317    } else {
318        av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n");
319    }
320
321    memcpy(pkt->data, read_data, read_length);
322    pa_stream_drop(pd->stream);
323
324    pa_threaded_mainloop_unlock(pd->mainloop);
325    return 0;
326
327unlock_and_fail:
328    pa_threaded_mainloop_unlock(pd->mainloop);
329    return ret;
330}
331
332static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
333{
334    PulseData *s = h->priv_data;
335    return ff_pulse_audio_get_devices(device_list, s->server, 0);
336}
337
338#define OFFSET(a) offsetof(PulseData, a)
339#define D AV_OPT_FLAG_DECODING_PARAM
340
341static const AVOption options[] = {
342    { "server",        "set PulseAudio server",                             OFFSET(server),        AV_OPT_TYPE_STRING, {.str = NULL},     0, 0, D },
343    { "name",          "set application name",                              OFFSET(name),          AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT},  0, 0, D },
344    { "stream_name",   "set stream description",                            OFFSET(stream_name),   AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
345    { "sample_rate",   "set sample rate in Hz",                             OFFSET(sample_rate),   AV_OPT_TYPE_INT,    {.i64 = 48000},    1, INT_MAX, D },
346    { "channels",      "set number of audio channels",                      OFFSET(channels),      AV_OPT_TYPE_INT,    {.i64 = 2},        1, INT_MAX, D },
347    { "frame_size",    "set number of bytes per frame",                     OFFSET(frame_size),    AV_OPT_TYPE_INT,    {.i64 = 1024},     1, INT_MAX, D },
348    { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT,    {.i64 = -1},      -1, INT_MAX, D },
349    { NULL },
350};
351
352static const AVClass pulse_demuxer_class = {
353    .class_name     = "Pulse demuxer",
354    .item_name      = av_default_item_name,
355    .option         = options,
356    .version        = LIBAVUTIL_VERSION_INT,
357    .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
358};
359
360AVInputFormat ff_pulse_demuxer = {
361    .name           = "pulse",
362    .long_name      = NULL_IF_CONFIG_SMALL("Pulse audio input"),
363    .priv_data_size = sizeof(PulseData),
364    .read_header    = pulse_read_header,
365    .read_packet    = pulse_read_packet,
366    .read_close     = pulse_close,
367    .get_device_list = pulse_get_device_list,
368    .flags          = AVFMT_NOFILE,
369    .priv_class     = &pulse_demuxer_class,
370};
371