1 /*
2 * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
3 * Copyright (C) 2013 Collabora Ltd.
4 * Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
5 * Copyright (C) 2018 Centricular Ltd.
6 * Author: Nirbheek Chauhan <nirbheek@centricular.com>
7 *
8 * This library is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Library General Public
10 * License as published by the Free Software Foundation; either
11 * version 2 of the License, or (at your option) any later version.
12 *
13 * This library 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 * Library General Public License for more details.
17 *
18 * You should have received a copy of the GNU Library General Public
19 * License along with this library; if not, write to the
20 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
21 * Boston, MA 02110-1301, USA.
22 */
23
24 /**
25 * SECTION:element-wasapisink
26 * @title: wasapisink
27 *
28 * Provides audio playback using the Windows Audio Session API available with
29 * Vista and newer.
30 *
31 * ## Example pipelines
32 * |[
33 * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink
34 * ]| Generate 20 ms buffers and render to the default audio device.
35 *
36 * |[
37 * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink low-latency=true
38 * ]| Same as above, but with the minimum possible latency
39 *
40 */
41 #ifdef HAVE_CONFIG_H
42 # include <config.h>
43 #endif
44
45 #include "gstwasapisink.h"
46
47 #include <avrt.h>
48
49 GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
50 #define GST_CAT_DEFAULT gst_wasapi_sink_debug
51
52 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
53 GST_PAD_SINK,
54 GST_PAD_ALWAYS,
55 GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS));
56
57 #define DEFAULT_ROLE GST_WASAPI_DEVICE_ROLE_CONSOLE
58 #define DEFAULT_MUTE FALSE
59 #define DEFAULT_EXCLUSIVE FALSE
60 #define DEFAULT_LOW_LATENCY FALSE
61 #define DEFAULT_AUDIOCLIENT3 TRUE
62
63 enum
64 {
65 PROP_0,
66 PROP_ROLE,
67 PROP_MUTE,
68 PROP_DEVICE,
69 PROP_EXCLUSIVE,
70 PROP_LOW_LATENCY,
71 PROP_AUDIOCLIENT3
72 };
73
74 static void gst_wasapi_sink_dispose (GObject * object);
75 static void gst_wasapi_sink_finalize (GObject * object);
76 static void gst_wasapi_sink_set_property (GObject * object, guint prop_id,
77 const GValue * value, GParamSpec * pspec);
78 static void gst_wasapi_sink_get_property (GObject * object, guint prop_id,
79 GValue * value, GParamSpec * pspec);
80
81 static GstCaps *gst_wasapi_sink_get_caps (GstBaseSink * bsink,
82 GstCaps * filter);
83
84 static gboolean gst_wasapi_sink_prepare (GstAudioSink * asink,
85 GstAudioRingBufferSpec * spec);
86 static gboolean gst_wasapi_sink_unprepare (GstAudioSink * asink);
87 static gboolean gst_wasapi_sink_open (GstAudioSink * asink);
88 static gboolean gst_wasapi_sink_close (GstAudioSink * asink);
89 static gint gst_wasapi_sink_write (GstAudioSink * asink,
90 gpointer data, guint length);
91 static guint gst_wasapi_sink_delay (GstAudioSink * asink);
92 static void gst_wasapi_sink_reset (GstAudioSink * asink);
93
94 #define gst_wasapi_sink_parent_class parent_class
95 G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_SINK);
96
97 static void
gst_wasapi_sink_class_init(GstWasapiSinkClass * klass)98 gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
99 {
100 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
101 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
102 GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
103 GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass);
104
105 gobject_class->dispose = gst_wasapi_sink_dispose;
106 gobject_class->finalize = gst_wasapi_sink_finalize;
107 gobject_class->set_property = gst_wasapi_sink_set_property;
108 gobject_class->get_property = gst_wasapi_sink_get_property;
109
110 g_object_class_install_property (gobject_class,
111 PROP_ROLE,
112 g_param_spec_enum ("role", "Role",
113 "Role of the device: communications, multimedia, etc",
114 GST_WASAPI_DEVICE_TYPE_ROLE, DEFAULT_ROLE, G_PARAM_READWRITE |
115 G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_READY));
116
117 g_object_class_install_property (gobject_class,
118 PROP_MUTE,
119 g_param_spec_boolean ("mute", "Mute", "Mute state of this stream",
120 DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
121 GST_PARAM_MUTABLE_PLAYING));
122
123 g_object_class_install_property (gobject_class,
124 PROP_DEVICE,
125 g_param_spec_string ("device", "Device",
126 "WASAPI playback device as a GUID string",
127 NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
128
129 g_object_class_install_property (gobject_class,
130 PROP_EXCLUSIVE,
131 g_param_spec_boolean ("exclusive", "Exclusive mode",
132 "Open the device in exclusive mode",
133 DEFAULT_EXCLUSIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
134
135 g_object_class_install_property (gobject_class,
136 PROP_LOW_LATENCY,
137 g_param_spec_boolean ("low-latency", "Low latency",
138 "Optimize all settings for lowest latency. Always safe to enable.",
139 DEFAULT_LOW_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
140
141 g_object_class_install_property (gobject_class,
142 PROP_AUDIOCLIENT3,
143 g_param_spec_boolean ("use-audioclient3", "Use the AudioClient3 API",
144 "Use the Windows 10 AudioClient3 API when available and if the "
145 "low-latency property is set to TRUE",
146 DEFAULT_AUDIOCLIENT3, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
147
148 gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
149 gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
150 "Sink/Audio/Hardware",
151 "Stream audio to an audio capture device through WASAPI",
152 "Nirbheek Chauhan <nirbheek@centricular.com>, "
153 "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
154
155 gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_sink_get_caps);
156
157 gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_prepare);
158 gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_unprepare);
159 gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_sink_open);
160 gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_sink_close);
161 gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_wasapi_sink_write);
162 gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_sink_delay);
163 gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_sink_reset);
164
165 GST_DEBUG_CATEGORY_INIT (gst_wasapi_sink_debug, "wasapisink",
166 0, "Windows audio session API sink");
167
168 gst_type_mark_as_plugin_api (GST_WASAPI_DEVICE_TYPE_ROLE, 0);
169 }
170
171 static void
gst_wasapi_sink_init(GstWasapiSink * self)172 gst_wasapi_sink_init (GstWasapiSink * self)
173 {
174 self->role = DEFAULT_ROLE;
175 self->mute = DEFAULT_MUTE;
176 self->sharemode = AUDCLNT_SHAREMODE_SHARED;
177 self->low_latency = DEFAULT_LOW_LATENCY;
178 self->try_audioclient3 = DEFAULT_AUDIOCLIENT3;
179 self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
180 self->cancellable = CreateEvent (NULL, TRUE, FALSE, NULL);
181 self->client_needs_restart = FALSE;
182
183 self->enumerator = gst_mm_device_enumerator_new ();
184 }
185
186 static void
gst_wasapi_sink_dispose(GObject * object)187 gst_wasapi_sink_dispose (GObject * object)
188 {
189 GstWasapiSink *self = GST_WASAPI_SINK (object);
190
191 if (self->event_handle != NULL) {
192 CloseHandle (self->event_handle);
193 self->event_handle = NULL;
194 }
195
196 if (self->cancellable != NULL) {
197 CloseHandle (self->cancellable);
198 self->cancellable = NULL;
199 }
200
201 if (self->client != NULL) {
202 IUnknown_Release (self->client);
203 self->client = NULL;
204 }
205
206 if (self->render_client != NULL) {
207 IUnknown_Release (self->render_client);
208 self->render_client = NULL;
209 }
210
211 gst_clear_object (&self->enumerator);
212
213 G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->dispose (object);
214 }
215
216 static void
gst_wasapi_sink_finalize(GObject * object)217 gst_wasapi_sink_finalize (GObject * object)
218 {
219 GstWasapiSink *self = GST_WASAPI_SINK (object);
220
221 CoTaskMemFree (self->mix_format);
222 self->mix_format = NULL;
223
224 if (self->cached_caps != NULL) {
225 gst_caps_unref (self->cached_caps);
226 self->cached_caps = NULL;
227 }
228
229 g_clear_pointer (&self->positions, g_free);
230 g_clear_pointer (&self->device_strid, g_free);
231 self->mute = FALSE;
232
233 G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object);
234 }
235
236 static void
gst_wasapi_sink_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)237 gst_wasapi_sink_set_property (GObject * object, guint prop_id,
238 const GValue * value, GParamSpec * pspec)
239 {
240 GstWasapiSink *self = GST_WASAPI_SINK (object);
241
242 switch (prop_id) {
243 case PROP_ROLE:
244 self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value));
245 break;
246 case PROP_MUTE:
247 self->mute = g_value_get_boolean (value);
248 break;
249 case PROP_DEVICE:
250 {
251 const gchar *device = g_value_get_string (value);
252 g_free (self->device_strid);
253 self->device_strid =
254 device ? g_utf8_to_utf16 (device, -1, NULL, NULL, NULL) : NULL;
255 break;
256 }
257 case PROP_EXCLUSIVE:
258 self->sharemode = g_value_get_boolean (value)
259 ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED;
260 break;
261 case PROP_LOW_LATENCY:
262 self->low_latency = g_value_get_boolean (value);
263 break;
264 case PROP_AUDIOCLIENT3:
265 self->try_audioclient3 = g_value_get_boolean (value);
266 break;
267 default:
268 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
269 break;
270 }
271 }
272
273 static void
gst_wasapi_sink_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)274 gst_wasapi_sink_get_property (GObject * object, guint prop_id,
275 GValue * value, GParamSpec * pspec)
276 {
277 GstWasapiSink *self = GST_WASAPI_SINK (object);
278
279 switch (prop_id) {
280 case PROP_ROLE:
281 g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role));
282 break;
283 case PROP_MUTE:
284 g_value_set_boolean (value, self->mute);
285 break;
286 case PROP_DEVICE:
287 g_value_take_string (value, self->device_strid ?
288 g_utf16_to_utf8 (self->device_strid, -1, NULL, NULL, NULL) : NULL);
289 break;
290 case PROP_EXCLUSIVE:
291 g_value_set_boolean (value,
292 self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE);
293 break;
294 case PROP_LOW_LATENCY:
295 g_value_set_boolean (value, self->low_latency);
296 break;
297 case PROP_AUDIOCLIENT3:
298 g_value_set_boolean (value, self->try_audioclient3);
299 break;
300 default:
301 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
302 break;
303 }
304 }
305
306 static gboolean
gst_wasapi_sink_can_audioclient3(GstWasapiSink * self)307 gst_wasapi_sink_can_audioclient3 (GstWasapiSink * self)
308 {
309 /* AudioClient3 API only makes sense in shared mode */
310 if (self->sharemode != AUDCLNT_SHAREMODE_SHARED)
311 return FALSE;
312
313 if (!self->try_audioclient3) {
314 GST_INFO_OBJECT (self, "AudioClient3 disabled by user");
315 return FALSE;
316 }
317
318 if (!gst_wasapi_util_have_audioclient3 ()) {
319 GST_INFO_OBJECT (self, "AudioClient3 not available on this OS");
320 return FALSE;
321 }
322
323 /* Only use audioclient3 when low-latency is requested because otherwise
324 * very slow machines and VMs with 1 CPU allocated will get glitches:
325 * https://bugzilla.gnome.org/show_bug.cgi?id=794497 */
326 if (!self->low_latency) {
327 GST_INFO_OBJECT (self, "AudioClient3 disabled because low-latency mode "
328 "was not requested");
329 return FALSE;
330 }
331
332 return TRUE;
333 }
334
335 static GstCaps *
gst_wasapi_sink_get_caps(GstBaseSink * bsink,GstCaps * filter)336 gst_wasapi_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
337 {
338 GstWasapiSink *self = GST_WASAPI_SINK (bsink);
339 WAVEFORMATEX *format = NULL;
340 GstCaps *caps = NULL;
341
342 GST_DEBUG_OBJECT (self, "entering get caps");
343
344 if (self->cached_caps) {
345 caps = gst_caps_ref (self->cached_caps);
346 } else {
347 GstCaps *template_caps;
348 gboolean ret;
349
350 template_caps = gst_pad_get_pad_template_caps (bsink->sinkpad);
351
352 if (!self->client) {
353 caps = template_caps;
354 goto out;
355 }
356
357 ret = gst_wasapi_util_get_device_format (GST_ELEMENT (self),
358 self->sharemode, self->device, self->client, &format);
359 if (!ret) {
360 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
361 ("failed to detect format"));
362 gst_caps_unref (template_caps);
363 return NULL;
364 }
365
366 gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
367 template_caps, &caps, &self->positions);
368 if (caps == NULL) {
369 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("unknown format"));
370 gst_caps_unref (template_caps);
371 return NULL;
372 }
373
374 {
375 gchar *pos_str = gst_audio_channel_positions_to_string (self->positions,
376 format->nChannels);
377 GST_INFO_OBJECT (self, "positions are: %s", pos_str);
378 g_free (pos_str);
379 }
380
381 self->mix_format = format;
382 gst_caps_replace (&self->cached_caps, caps);
383 gst_caps_unref (template_caps);
384 }
385
386 if (filter) {
387 GstCaps *filtered =
388 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
389 gst_caps_unref (caps);
390 caps = filtered;
391 }
392
393 out:
394 GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps);
395 return caps;
396 }
397
398 static gboolean
gst_wasapi_sink_open(GstAudioSink * asink)399 gst_wasapi_sink_open (GstAudioSink * asink)
400 {
401 GstWasapiSink *self = GST_WASAPI_SINK (asink);
402 gboolean res = FALSE;
403 IMMDevice *device = NULL;
404 IAudioClient *client = NULL;
405
406 GST_DEBUG_OBJECT (self, "opening device");
407
408 if (self->client)
409 return TRUE;
410
411 /* FIXME: Switching the default device does not switch the stream to it,
412 * even if the old device was unplugged. We need to handle this somehow.
413 * For example, perhaps we should automatically switch to the new device if
414 * the default device is changed and a device isn't explicitly selected. */
415 if (!gst_wasapi_util_get_device (self->enumerator, eRender,
416 self->role, self->device_strid, &device)
417 || !gst_wasapi_util_get_audio_client (GST_ELEMENT (self),
418 device, &client)) {
419 if (!self->device_strid)
420 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
421 ("Failed to get default device"));
422 else
423 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
424 ("Failed to open device %S", self->device_strid));
425 goto beach;
426 }
427
428 self->client = client;
429 self->device = device;
430 res = TRUE;
431
432 beach:
433
434 return res;
435 }
436
437 static gboolean
gst_wasapi_sink_close(GstAudioSink * asink)438 gst_wasapi_sink_close (GstAudioSink * asink)
439 {
440 GstWasapiSink *self = GST_WASAPI_SINK (asink);
441
442 if (self->device != NULL) {
443 IUnknown_Release (self->device);
444 self->device = NULL;
445 }
446
447 if (self->client != NULL) {
448 IUnknown_Release (self->client);
449 self->client = NULL;
450 }
451
452 return TRUE;
453 }
454
455 /* Get the empty space in the buffer that we have to write to */
456 static gint
gst_wasapi_sink_get_can_frames(GstWasapiSink * self)457 gst_wasapi_sink_get_can_frames (GstWasapiSink * self)
458 {
459 HRESULT hr;
460 guint n_frames_padding;
461
462 /* There is no padding in exclusive mode since there is no ringbuffer */
463 if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) {
464 GST_DEBUG_OBJECT (self, "exclusive mode, can write: %i",
465 self->buffer_frame_count);
466 return self->buffer_frame_count;
467 }
468
469 /* Frames the card hasn't rendered yet */
470 hr = IAudioClient_GetCurrentPadding (self->client, &n_frames_padding);
471 HR_FAILED_ELEMENT_ERROR_RET (hr, IAudioClient::GetCurrentPadding, self, -1);
472
473 GST_DEBUG_OBJECT (self, "%i unread frames (padding)", n_frames_padding);
474
475 /* We can write out these many frames */
476 return self->buffer_frame_count - n_frames_padding;
477 }
478
479 static gboolean
gst_wasapi_sink_prepare(GstAudioSink * asink,GstAudioRingBufferSpec * spec)480 gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
481 {
482 GstWasapiSink *self = GST_WASAPI_SINK (asink);
483 gboolean res = FALSE;
484 REFERENCE_TIME latency_rt;
485 guint bpf, rate, devicep_frames;
486 HRESULT hr;
487
488 if (!self->client) {
489 GST_DEBUG_OBJECT (self, "no IAudioClient, creating a new one");
490 if (!gst_wasapi_util_get_audio_client (GST_ELEMENT (self),
491 self->device, &self->client))
492 goto beach;
493 }
494
495 if (gst_wasapi_sink_can_audioclient3 (self)) {
496 if (!gst_wasapi_util_initialize_audioclient3 (GST_ELEMENT (self), spec,
497 (IAudioClient3 *) self->client, self->mix_format, self->low_latency,
498 FALSE, &devicep_frames))
499 goto beach;
500 } else {
501 if (!gst_wasapi_util_initialize_audioclient (GST_ELEMENT (self), spec,
502 self->client, self->mix_format, self->sharemode, self->low_latency,
503 FALSE, &devicep_frames))
504 goto beach;
505 }
506
507 bpf = GST_AUDIO_INFO_BPF (&spec->info);
508 rate = GST_AUDIO_INFO_RATE (&spec->info);
509
510 /* Total size of the allocated buffer that we will write to */
511 hr = IAudioClient_GetBufferSize (self->client, &self->buffer_frame_count);
512 HR_FAILED_GOTO (hr, IAudioClient::GetBufferSize, beach);
513
514 GST_INFO_OBJECT (self, "buffer size is %i frames, device period is %i "
515 "frames, bpf is %i bytes, rate is %i Hz", self->buffer_frame_count,
516 devicep_frames, bpf, rate);
517
518 /* Actual latency-time/buffer-time will be different now */
519 spec->segsize = devicep_frames * bpf;
520
521 /* We need a minimum of 2 segments to ensure glitch-free playback */
522 spec->segtotal = MAX (self->buffer_frame_count * bpf / spec->segsize, 2);
523
524 GST_INFO_OBJECT (self, "segsize is %i, segtotal is %i", spec->segsize,
525 spec->segtotal);
526
527 /* Get latency for logging */
528 hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
529 HR_FAILED_GOTO (hr, IAudioClient::GetStreamLatency, beach);
530
531 GST_INFO_OBJECT (self, "wasapi stream latency: %" G_GINT64_FORMAT " (%"
532 G_GINT64_FORMAT "ms)", latency_rt, latency_rt / 10000);
533
534 /* Set the event handler which will trigger writes */
535 hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
536 HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach);
537
538 /* Get render sink client and start it up */
539 if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self), self->client,
540 &self->render_client)) {
541 goto beach;
542 }
543
544 GST_INFO_OBJECT (self, "got render client");
545
546 /* To avoid start-up glitches, before starting the streaming, we fill the
547 * buffer with silence as recommended by the documentation:
548 * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370879%28v=vs.85%29.aspx */
549 {
550 gint n_frames, len;
551 gint16 *dst = NULL;
552
553 n_frames = gst_wasapi_sink_get_can_frames (self);
554 if (n_frames < 1) {
555 GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
556 ("should have more than %i frames to write", n_frames));
557 goto beach;
558 }
559
560 len = n_frames * self->mix_format->nBlockAlign;
561
562 hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames,
563 (BYTE **) & dst);
564 HR_FAILED_GOTO (hr, IAudioRenderClient::GetBuffer, beach);
565
566 GST_DEBUG_OBJECT (self, "pre-wrote %i bytes of silence", len);
567
568 hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames,
569 AUDCLNT_BUFFERFLAGS_SILENT);
570 HR_FAILED_GOTO (hr, IAudioRenderClient::ReleaseBuffer, beach);
571 }
572
573 hr = IAudioClient_Start (self->client);
574 HR_FAILED_GOTO (hr, IAudioClient::Start, beach);
575 self->client_needs_restart = FALSE;
576
577 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK
578 (self)->ringbuffer, self->positions);
579
580 res = TRUE;
581
582 /* reset cancellable event handle */
583 ResetEvent (self->cancellable);
584
585 beach:
586 /* unprepare() is not called if prepare() fails, but we want it to be, so call
587 * it manually when needed */
588 if (!res)
589 gst_wasapi_sink_unprepare (asink);
590
591 return res;
592 }
593
594 static gboolean
gst_wasapi_sink_unprepare(GstAudioSink * asink)595 gst_wasapi_sink_unprepare (GstAudioSink * asink)
596 {
597 GstWasapiSink *self = GST_WASAPI_SINK (asink);
598
599 if (self->client != NULL) {
600 IUnknown_Release (self->client);
601 self->client = NULL;
602 }
603
604 if (self->render_client != NULL) {
605 IUnknown_Release (self->render_client);
606 self->render_client = NULL;
607 }
608
609 return TRUE;
610 }
611
612 static gint
gst_wasapi_sink_write(GstAudioSink * asink,gpointer data,guint length)613 gst_wasapi_sink_write (GstAudioSink * asink, gpointer data, guint length)
614 {
615 GstWasapiSink *self = GST_WASAPI_SINK (asink);
616 HRESULT hr;
617 gint16 *dst = NULL;
618 DWORD dwWaitResult;
619 guint can_frames, have_frames, n_frames, write_len, written_len = 0;
620 HANDLE event_handle[2];
621
622 event_handle[0] = self->event_handle;
623 event_handle[1] = self->cancellable;
624
625 GST_OBJECT_LOCK (self);
626 if (self->client_needs_restart) {
627 hr = IAudioClient_Start (self->client);
628 HR_FAILED_ELEMENT_ERROR_AND (hr, IAudioClient::Start, self,
629 GST_OBJECT_UNLOCK (self); goto err);
630 self->client_needs_restart = FALSE;
631 ResetEvent (self->cancellable);
632 }
633 GST_OBJECT_UNLOCK (self);
634
635 /* We have N frames to be written out */
636 have_frames = length / (self->mix_format->nBlockAlign);
637
638 if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) {
639 /* In exclusive mode we have to wait always */
640 dwWaitResult = WaitForMultipleObjects (2, event_handle, FALSE, INFINITE);
641 if (dwWaitResult != WAIT_OBJECT_0 && dwWaitResult != WAIT_OBJECT_0 + 1) {
642 GST_ERROR_OBJECT (self, "Error waiting for event handle: %x",
643 (guint) dwWaitResult);
644 goto err;
645 }
646
647 /* ::reset was requested */
648 if (dwWaitResult == WAIT_OBJECT_0 + 1) {
649 GST_DEBUG_OBJECT (self, "operation was cancelled");
650 return -1;
651 }
652
653 can_frames = gst_wasapi_sink_get_can_frames (self);
654 if (can_frames < 0) {
655 GST_ERROR_OBJECT (self, "Error getting frames to write to");
656 goto err;
657 }
658 /* In exclusive mode we need to fill the whole buffer in one go or
659 * GetBuffer will error out */
660 if (can_frames != have_frames) {
661 GST_ERROR_OBJECT (self,
662 "Need at %i frames to write for exclusive mode, but got %i",
663 can_frames, have_frames);
664 goto err;
665 }
666 } else {
667 /* In shared mode we can write parts of the buffer, so only wait
668 * in case we can't write anything */
669 can_frames = gst_wasapi_sink_get_can_frames (self);
670 if (can_frames < 0) {
671 GST_ERROR_OBJECT (self, "Error getting frames to write to");
672 goto err;
673 }
674
675 if (can_frames == 0) {
676 dwWaitResult = WaitForMultipleObjects (2, event_handle, FALSE, INFINITE);
677 if (dwWaitResult != WAIT_OBJECT_0 && dwWaitResult != WAIT_OBJECT_0 + 1) {
678 GST_ERROR_OBJECT (self, "Error waiting for event handle: %x",
679 (guint) dwWaitResult);
680 goto err;
681 }
682
683 /* ::reset was requested */
684 if (dwWaitResult == WAIT_OBJECT_0 + 1) {
685 GST_DEBUG_OBJECT (self, "operation was cancelled");
686 return -1;
687 }
688
689 can_frames = gst_wasapi_sink_get_can_frames (self);
690 if (can_frames < 0) {
691 GST_ERROR_OBJECT (self, "Error getting frames to write to");
692 goto err;
693 }
694 }
695 }
696
697 /* We will write out these many frames, and this much length */
698 n_frames = MIN (can_frames, have_frames);
699 write_len = n_frames * self->mix_format->nBlockAlign;
700
701 GST_DEBUG_OBJECT (self, "total: %i, have_frames: %i (%i bytes), "
702 "can_frames: %i, will write: %i (%i bytes)", self->buffer_frame_count,
703 have_frames, length, can_frames, n_frames, write_len);
704
705 hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames,
706 (BYTE **) & dst);
707 HR_FAILED_ELEMENT_ERROR_AND (hr, IAudioRenderClient::GetBuffer, self,
708 goto err);
709
710 memcpy (dst, data, write_len);
711
712 hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames,
713 self->mute ? AUDCLNT_BUFFERFLAGS_SILENT : 0);
714 HR_FAILED_ELEMENT_ERROR_AND (hr, IAudioRenderClient::ReleaseBuffer, self,
715 goto err);
716
717 written_len = write_len;
718
719 out:
720 return written_len;
721
722 err:
723 written_len = -1;
724 goto out;
725 }
726
727 static guint
gst_wasapi_sink_delay(GstAudioSink * asink)728 gst_wasapi_sink_delay (GstAudioSink * asink)
729 {
730 GstWasapiSink *self = GST_WASAPI_SINK (asink);
731 guint delay = 0;
732 HRESULT hr;
733
734 hr = IAudioClient_GetCurrentPadding (self->client, &delay);
735 HR_FAILED_RET (hr, IAudioClient::GetCurrentPadding, 0);
736
737 return delay;
738 }
739
740 static void
gst_wasapi_sink_reset(GstAudioSink * asink)741 gst_wasapi_sink_reset (GstAudioSink * asink)
742 {
743 GstWasapiSink *self = GST_WASAPI_SINK (asink);
744 HRESULT hr;
745
746 GST_INFO_OBJECT (self, "reset called");
747
748 if (!self->client)
749 return;
750
751 SetEvent (self->cancellable);
752
753 GST_OBJECT_LOCK (self);
754 hr = IAudioClient_Stop (self->client);
755 HR_FAILED_AND (hr, IAudioClient::Stop, goto err);
756
757 hr = IAudioClient_Reset (self->client);
758 HR_FAILED_AND (hr, IAudioClient::Reset, goto err);
759
760 err:
761 self->client_needs_restart = TRUE;
762 GST_OBJECT_UNLOCK (self);
763 }
764