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>
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.
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.
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.
25 * SECTION:element-wasapisink
28 * Provides audio playback using the Windows Audio Session API available with
31 * ## Example pipelines
33 * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink
34 * ]| Generate 20 ms buffers and render to the default audio device.
37 * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink low-latency=true
38 * ]| Same as above, but with the minimum possible latency
45 #include "gstwasapisink.h"
49 GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
50 #define GST_CAT_DEFAULT gst_wasapi_sink_debug
52 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
55 GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS));
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
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);
81 static GstCaps *gst_wasapi_sink_get_caps (GstBaseSink * bsink,
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);
94 #define gst_wasapi_sink_parent_class parent_class
95 G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_SINK);
98 gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
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);
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;
110 g_object_class_install_property (gobject_class,
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));
117 g_object_class_install_property (gobject_class,
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));
123 g_object_class_install_property (gobject_class,
125 g_param_spec_string ("device", "Device",
126 "WASAPI playback device as a GUID string",
127 NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
129 g_object_class_install_property (gobject_class,
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));
135 g_object_class_install_property (gobject_class,
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));
141 g_object_class_install_property (gobject_class,
143 g_param_spec_boolean ("use-audioclient3", "Use the AudioClient3 API",
144 "Use the Windows 10 AudioClient3 API when available",
145 DEFAULT_AUDIOCLIENT3, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
147 gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
148 gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
150 "Stream audio to an audio capture device through WASAPI",
151 "Nirbheek Chauhan <nirbheek@centricular.com>, "
152 "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
154 gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_sink_get_caps);
156 gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_prepare);
157 gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_unprepare);
158 gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_sink_open);
159 gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_sink_close);
160 gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_wasapi_sink_write);
161 gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_sink_delay);
162 gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_sink_reset);
164 GST_DEBUG_CATEGORY_INIT (gst_wasapi_sink_debug, "wasapisink",
165 0, "Windows audio session API sink");
169 gst_wasapi_sink_init (GstWasapiSink * self)
171 self->role = DEFAULT_ROLE;
172 self->mute = DEFAULT_MUTE;
173 self->sharemode = AUDCLNT_SHAREMODE_SHARED;
174 self->low_latency = DEFAULT_LOW_LATENCY;
175 self->try_audioclient3 = DEFAULT_AUDIOCLIENT3;
176 self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
182 gst_wasapi_sink_dispose (GObject * object)
184 GstWasapiSink *self = GST_WASAPI_SINK (object);
186 if (self->event_handle != NULL) {
187 CloseHandle (self->event_handle);
188 self->event_handle = NULL;
191 if (self->client != NULL) {
192 IUnknown_Release (self->client);
196 if (self->render_client != NULL) {
197 IUnknown_Release (self->render_client);
198 self->render_client = NULL;
201 G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->dispose (object);
205 gst_wasapi_sink_finalize (GObject * object)
207 GstWasapiSink *self = GST_WASAPI_SINK (object);
209 g_clear_pointer (&self->mix_format, CoTaskMemFree);
213 if (self->cached_caps != NULL) {
214 gst_caps_unref (self->cached_caps);
215 self->cached_caps = NULL;
218 g_clear_pointer (&self->positions, g_free);
219 g_clear_pointer (&self->device_strid, g_free);
222 G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object);
226 gst_wasapi_sink_set_property (GObject * object, guint prop_id,
227 const GValue * value, GParamSpec * pspec)
229 GstWasapiSink *self = GST_WASAPI_SINK (object);
233 self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value));
236 self->mute = g_value_get_boolean (value);
240 const gchar *device = g_value_get_string (value);
241 g_free (self->device_strid);
243 device ? g_utf8_to_utf16 (device, -1, NULL, NULL, NULL) : NULL;
247 self->sharemode = g_value_get_boolean (value)
248 ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED;
250 case PROP_LOW_LATENCY:
251 self->low_latency = g_value_get_boolean (value);
253 case PROP_AUDIOCLIENT3:
254 self->try_audioclient3 = g_value_get_boolean (value);
257 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
263 gst_wasapi_sink_get_property (GObject * object, guint prop_id,
264 GValue * value, GParamSpec * pspec)
266 GstWasapiSink *self = GST_WASAPI_SINK (object);
270 g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role));
273 g_value_set_boolean (value, self->mute);
276 g_value_take_string (value, self->device_strid ?
277 g_utf16_to_utf8 (self->device_strid, -1, NULL, NULL, NULL) : NULL);
280 g_value_set_boolean (value,
281 self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE);
283 case PROP_LOW_LATENCY:
284 g_value_set_boolean (value, self->low_latency);
286 case PROP_AUDIOCLIENT3:
287 g_value_set_boolean (value, self->try_audioclient3);
290 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
296 gst_wasapi_sink_can_audioclient3 (GstWasapiSink * self)
298 if (self->sharemode == AUDCLNT_SHAREMODE_SHARED &&
299 self->try_audioclient3 && gst_wasapi_util_have_audioclient3 ())
305 gst_wasapi_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
307 GstWasapiSink *self = GST_WASAPI_SINK (bsink);
308 WAVEFORMATEX *format = NULL;
309 GstCaps *caps = NULL;
311 GST_DEBUG_OBJECT (self, "entering get caps");
313 if (self->cached_caps) {
314 caps = gst_caps_ref (self->cached_caps);
316 GstCaps *template_caps;
319 template_caps = gst_pad_get_pad_template_caps (bsink->sinkpad);
322 gst_wasapi_sink_open (GST_AUDIO_SINK (bsink));
324 ret = gst_wasapi_util_get_device_format (GST_ELEMENT (self),
325 self->sharemode, self->device, self->client, &format);
327 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
328 ("failed to detect format"));
332 gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
333 template_caps, &caps, &self->positions);
335 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("unknown format"));
340 gchar *pos_str = gst_audio_channel_positions_to_string (self->positions,
342 GST_INFO_OBJECT (self, "positions are: %s", pos_str);
346 self->mix_format = format;
347 gst_caps_replace (&self->cached_caps, caps);
348 gst_caps_unref (template_caps);
353 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
354 gst_caps_unref (caps);
358 GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps);
365 gst_wasapi_sink_open (GstAudioSink * asink)
367 GstWasapiSink *self = GST_WASAPI_SINK (asink);
368 gboolean res = FALSE;
369 IMMDevice *device = NULL;
370 IAudioClient *client = NULL;
372 GST_DEBUG_OBJECT (self, "opening device");
377 /* FIXME: Switching the default device does not switch the stream to it,
378 * even if the old device was unplugged. We need to handle this somehow.
379 * For example, perhaps we should automatically switch to the new device if
380 * the default device is changed and a device isn't explicitly selected. */
381 if (!gst_wasapi_util_get_device_client (GST_ELEMENT (self), eRender,
382 self->role, self->device_strid, &device, &client)) {
383 if (!self->device_strid)
384 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
385 ("Failed to get default device"));
387 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
388 ("Failed to open device %S", self->device_strid));
392 self->client = client;
393 self->device = device;
402 gst_wasapi_sink_close (GstAudioSink * asink)
404 GstWasapiSink *self = GST_WASAPI_SINK (asink);
406 if (self->device != NULL) {
407 IUnknown_Release (self->device);
411 if (self->client != NULL) {
412 IUnknown_Release (self->client);
419 /* Get the empty space in the buffer that we have to write to */
421 gst_wasapi_sink_get_can_frames (GstWasapiSink * self)
424 guint n_frames_padding;
426 /* There is no padding in exclusive mode since there is no ringbuffer */
427 if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) {
428 GST_DEBUG_OBJECT (self, "exclusive mode, can write: %i",
429 self->buffer_frame_count);
430 return self->buffer_frame_count;
433 /* Frames the card hasn't rendered yet */
434 hr = IAudioClient_GetCurrentPadding (self->client, &n_frames_padding);
435 HR_FAILED_RET (hr, IAudioClient::GetCurrentPadding, -1);
437 GST_DEBUG_OBJECT (self, "%i unread frames (padding)", n_frames_padding);
439 /* We can write out these many frames */
440 return self->buffer_frame_count - n_frames_padding;
444 gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
446 GstWasapiSink *self = GST_WASAPI_SINK (asink);
447 gboolean res = FALSE;
448 REFERENCE_TIME latency_rt;
449 guint bpf, rate, devicep_frames;
452 if (gst_wasapi_sink_can_audioclient3 (self)) {
453 if (!gst_wasapi_util_initialize_audioclient3 (GST_ELEMENT (self), spec,
454 (IAudioClient3 *) self->client, self->mix_format, self->low_latency,
455 FALSE, &devicep_frames))
458 if (!gst_wasapi_util_initialize_audioclient (GST_ELEMENT (self), spec,
459 self->client, self->mix_format, self->sharemode, self->low_latency,
460 FALSE, &devicep_frames))
464 bpf = GST_AUDIO_INFO_BPF (&spec->info);
465 rate = GST_AUDIO_INFO_RATE (&spec->info);
467 /* Total size of the allocated buffer that we will write to */
468 hr = IAudioClient_GetBufferSize (self->client, &self->buffer_frame_count);
469 HR_FAILED_GOTO (hr, IAudioClient::GetBufferSize, beach);
471 GST_INFO_OBJECT (self, "buffer size is %i frames, device period is %i "
472 "frames, bpf is %i bytes, rate is %i Hz", self->buffer_frame_count,
473 devicep_frames, bpf, rate);
475 /* Actual latency-time/buffer-time will be different now */
476 spec->segsize = devicep_frames * bpf;
478 /* We need a minimum of 2 segments to ensure glitch-free playback */
479 spec->segtotal = MAX (self->buffer_frame_count * bpf / spec->segsize, 2);
481 GST_INFO_OBJECT (self, "segsize is %i, segtotal is %i", spec->segsize,
484 /* Get latency for logging */
485 hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
486 HR_FAILED_GOTO (hr, IAudioClient::GetStreamLatency, beach);
488 GST_INFO_OBJECT (self, "wasapi stream latency: %" G_GINT64_FORMAT " (%"
489 G_GINT64_FORMAT "ms)", latency_rt, latency_rt / 10000);
491 /* Set the event handler which will trigger writes */
492 hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
493 HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach);
495 /* Get render sink client and start it up */
496 if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self), self->client,
497 &self->render_client)) {
501 GST_INFO_OBJECT (self, "got render client");
503 /* To avoid start-up glitches, before starting the streaming, we fill the
504 * buffer with silence as recommended by the documentation:
505 * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370879%28v=vs.85%29.aspx */
510 n_frames = gst_wasapi_sink_get_can_frames (self);
512 GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
513 ("should have more than %i frames to write", n_frames));
517 len = n_frames * self->mix_format->nBlockAlign;
519 hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames,
521 HR_FAILED_GOTO (hr, IAudioRenderClient::GetBuffer, beach);
523 GST_DEBUG_OBJECT (self, "pre-wrote %i bytes of silence", len);
525 hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames,
526 AUDCLNT_BUFFERFLAGS_SILENT);
527 HR_FAILED_GOTO (hr, IAudioRenderClient::ReleaseBuffer, beach);
530 hr = IAudioClient_Start (self->client);
531 HR_FAILED_GOTO (hr, IAudioClient::Start, beach);
533 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK
534 (self)->ringbuffer, self->positions);
536 /* Increase the thread priority to reduce glitches */
537 self->thread_priority_handle = gst_wasapi_util_set_thread_characteristics ();
542 /* unprepare() is not called if prepare() fails, but we want it to be, so call
543 * it manually when needed */
545 gst_wasapi_sink_unprepare (asink);
551 gst_wasapi_sink_unprepare (GstAudioSink * asink)
553 GstWasapiSink *self = GST_WASAPI_SINK (asink);
555 if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE &&
556 !gst_wasapi_sink_can_audioclient3 (self))
559 if (self->thread_priority_handle != NULL) {
560 gst_wasapi_util_revert_thread_characteristics
561 (self->thread_priority_handle);
562 self->thread_priority_handle = NULL;
565 if (self->client != NULL) {
566 IAudioClient_Stop (self->client);
569 if (self->render_client != NULL) {
570 IUnknown_Release (self->render_client);
571 self->render_client = NULL;
578 gst_wasapi_sink_write (GstAudioSink * asink, gpointer data, guint length)
580 GstWasapiSink *self = GST_WASAPI_SINK (asink);
583 guint pending = length;
585 while (pending > 0) {
586 guint can_frames, have_frames, n_frames, write_len;
588 WaitForSingleObject (self->event_handle, INFINITE);
590 /* We have N frames to be written out */
591 have_frames = pending / (self->mix_format->nBlockAlign);
592 /* We have can_frames space in the output buffer */
593 can_frames = gst_wasapi_sink_get_can_frames (self);
594 /* We will write out these many frames, and this much length */
595 n_frames = MIN (can_frames, have_frames);
596 write_len = n_frames * self->mix_format->nBlockAlign;
598 GST_DEBUG_OBJECT (self, "total: %i, have_frames: %i (%i bytes), "
599 "can_frames: %i, will write: %i (%i bytes)", self->buffer_frame_count,
600 have_frames, pending, can_frames, n_frames, write_len);
602 hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames,
604 HR_FAILED_AND (hr, IAudioRenderClient::GetBuffer, length = 0; goto beach);
606 memcpy (dst, data, write_len);
608 hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames,
609 self->mute ? AUDCLNT_BUFFERFLAGS_SILENT : 0);
610 HR_FAILED_AND (hr, IAudioRenderClient::ReleaseBuffer, length = 0;
613 pending -= write_len;
622 gst_wasapi_sink_delay (GstAudioSink * asink)
624 GstWasapiSink *self = GST_WASAPI_SINK (asink);
628 hr = IAudioClient_GetCurrentPadding (self->client, &delay);
629 HR_FAILED_RET (hr, IAudioClient::GetCurrentPadding, 0);
635 gst_wasapi_sink_reset (GstAudioSink * asink)
637 GstWasapiSink *self = GST_WASAPI_SINK (asink);
643 hr = IAudioClient_Stop (self->client);
644 HR_FAILED_RET (hr, IAudioClient::Stop,);
646 hr = IAudioClient_Reset (self->client);
647 HR_FAILED_RET (hr, IAudioClient::Reset,);