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.
41 #include "gstwasapisink.h"
45 GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
46 #define GST_CAT_DEFAULT gst_wasapi_sink_debug
48 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
51 GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS));
53 #define DEFAULT_ROLE GST_WASAPI_DEVICE_ROLE_CONSOLE
54 #define DEFAULT_MUTE FALSE
55 #define DEFAULT_EXCLUSIVE FALSE
56 #define DEFAULT_LOW_LATENCY FALSE
68 static void gst_wasapi_sink_dispose (GObject * object);
69 static void gst_wasapi_sink_finalize (GObject * object);
70 static void gst_wasapi_sink_set_property (GObject * object, guint prop_id,
71 const GValue * value, GParamSpec * pspec);
72 static void gst_wasapi_sink_get_property (GObject * object, guint prop_id,
73 GValue * value, GParamSpec * pspec);
75 static GstCaps *gst_wasapi_sink_get_caps (GstBaseSink * bsink,
78 static gboolean gst_wasapi_sink_prepare (GstAudioSink * asink,
79 GstAudioRingBufferSpec * spec);
80 static gboolean gst_wasapi_sink_unprepare (GstAudioSink * asink);
81 static gboolean gst_wasapi_sink_open (GstAudioSink * asink);
82 static gboolean gst_wasapi_sink_close (GstAudioSink * asink);
83 static gint gst_wasapi_sink_write (GstAudioSink * asink,
84 gpointer data, guint length);
85 static guint gst_wasapi_sink_delay (GstAudioSink * asink);
86 static void gst_wasapi_sink_reset (GstAudioSink * asink);
88 #define gst_wasapi_sink_parent_class parent_class
89 G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_SINK);
92 gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
94 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
95 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
96 GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
97 GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass);
99 gobject_class->dispose = gst_wasapi_sink_dispose;
100 gobject_class->finalize = gst_wasapi_sink_finalize;
101 gobject_class->set_property = gst_wasapi_sink_set_property;
102 gobject_class->get_property = gst_wasapi_sink_get_property;
104 g_object_class_install_property (gobject_class,
106 g_param_spec_enum ("role", "Role",
107 "Role of the device: communications, multimedia, etc",
108 GST_WASAPI_DEVICE_TYPE_ROLE, DEFAULT_ROLE, G_PARAM_READWRITE |
109 G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_READY));
111 g_object_class_install_property (gobject_class,
113 g_param_spec_boolean ("mute", "Mute", "Mute state of this stream",
114 DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
115 GST_PARAM_MUTABLE_PLAYING));
117 g_object_class_install_property (gobject_class,
119 g_param_spec_string ("device", "Device",
120 "WASAPI playback device as a GUID string",
121 NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
123 g_object_class_install_property (gobject_class,
125 g_param_spec_boolean ("exclusive", "Exclusive mode",
126 "Open the device in exclusive mode",
127 DEFAULT_EXCLUSIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
129 g_object_class_install_property (gobject_class,
131 g_param_spec_boolean ("low-latency", "Low latency",
132 "Optimize all settings for lowest latency",
133 DEFAULT_LOW_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
135 gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
136 gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
138 "Stream audio to an audio capture device through WASAPI",
139 "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
141 gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_sink_get_caps);
143 gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_prepare);
144 gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_unprepare);
145 gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_sink_open);
146 gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_sink_close);
147 gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_wasapi_sink_write);
148 gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_sink_delay);
149 gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_sink_reset);
151 GST_DEBUG_CATEGORY_INIT (gst_wasapi_sink_debug, "wasapisink",
152 0, "Windows audio session API sink");
156 gst_wasapi_sink_init (GstWasapiSink * self)
158 self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
164 gst_wasapi_sink_dispose (GObject * object)
166 GstWasapiSink *self = GST_WASAPI_SINK (object);
168 if (self->event_handle != NULL) {
169 CloseHandle (self->event_handle);
170 self->event_handle = NULL;
173 if (self->client != NULL) {
174 IUnknown_Release (self->client);
178 if (self->render_client != NULL) {
179 IUnknown_Release (self->render_client);
180 self->render_client = NULL;
183 G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->dispose (object);
187 gst_wasapi_sink_finalize (GObject * object)
189 GstWasapiSink *self = GST_WASAPI_SINK (object);
191 g_clear_pointer (&self->mix_format, CoTaskMemFree);
195 if (self->cached_caps != NULL) {
196 gst_caps_unref (self->cached_caps);
197 self->cached_caps = NULL;
200 g_clear_pointer (&self->positions, g_free);
201 g_clear_pointer (&self->device_strid, g_free);
204 G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object);
208 gst_wasapi_sink_set_property (GObject * object, guint prop_id,
209 const GValue * value, GParamSpec * pspec)
211 GstWasapiSink *self = GST_WASAPI_SINK (object);
215 self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value));
218 self->mute = g_value_get_boolean (value);
222 const gchar *device = g_value_get_string (value);
223 g_free (self->device_strid);
225 device ? g_utf8_to_utf16 (device, -1, NULL, NULL, NULL) : NULL;
229 self->sharemode = g_value_get_boolean (value)
230 ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED;
232 case PROP_LOW_LATENCY:
233 self->low_latency = g_value_get_boolean (value);
236 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
242 gst_wasapi_sink_get_property (GObject * object, guint prop_id,
243 GValue * value, GParamSpec * pspec)
245 GstWasapiSink *self = GST_WASAPI_SINK (object);
249 g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role));
252 g_value_set_boolean (value, self->mute);
255 g_value_take_string (value, self->device_strid ?
256 g_utf16_to_utf8 (self->device_strid, -1, NULL, NULL, NULL) : NULL);
259 g_value_set_boolean (value,
260 self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE);
262 case PROP_LOW_LATENCY:
263 g_value_set_boolean (value, self->low_latency);
266 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
272 gst_wasapi_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
274 GstWasapiSink *self = GST_WASAPI_SINK (bsink);
275 WAVEFORMATEX *format = NULL;
276 GstCaps *caps = NULL;
278 GST_DEBUG_OBJECT (self, "entering get caps");
280 if (self->cached_caps) {
281 caps = gst_caps_ref (self->cached_caps);
283 GstCaps *template_caps;
286 template_caps = gst_pad_get_pad_template_caps (bsink->sinkpad);
289 gst_wasapi_sink_open (GST_AUDIO_SINK (bsink));
291 ret = gst_wasapi_util_get_device_format (GST_ELEMENT (self),
292 self->sharemode, self->device, self->client, &format);
294 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
295 ("failed to detect format"));
299 gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
300 template_caps, &caps, &self->positions);
302 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("unknown format"));
307 gchar *pos_str = gst_audio_channel_positions_to_string (self->positions,
309 GST_INFO_OBJECT (self, "positions are: %s", pos_str);
313 self->mix_format = format;
314 gst_caps_replace (&self->cached_caps, caps);
315 gst_caps_unref (template_caps);
320 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
321 gst_caps_unref (caps);
325 GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps);
332 gst_wasapi_sink_open (GstAudioSink * asink)
334 GstWasapiSink *self = GST_WASAPI_SINK (asink);
335 gboolean res = FALSE;
336 IMMDevice *device = NULL;
337 IAudioClient *client = NULL;
339 GST_DEBUG_OBJECT (self, "opening device");
344 /* FIXME: Switching the default device does not switch the stream to it,
345 * even if the old device was unplugged. We need to handle this somehow.
346 * For example, perhaps we should automatically switch to the new device if
347 * the default device is changed and a device isn't explicitly selected. */
348 if (!gst_wasapi_util_get_device_client (GST_ELEMENT (self), FALSE,
349 self->role, self->device_strid, &device, &client)) {
350 if (!self->device_strid)
351 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
352 ("Failed to get default device"));
354 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
355 ("Failed to open device %S", self->device_strid));
359 self->client = client;
360 self->device = device;
369 gst_wasapi_sink_close (GstAudioSink * asink)
371 GstWasapiSink *self = GST_WASAPI_SINK (asink);
373 if (self->device != NULL) {
374 IUnknown_Release (self->device);
378 if (self->client != NULL) {
379 IUnknown_Release (self->client);
386 /* Get the empty space in the buffer that we have to write to */
388 gst_wasapi_sink_get_can_frames (GstWasapiSink * self)
391 guint n_frames_padding;
393 /* There is no padding in exclusive mode since there is no ringbuffer */
394 if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) {
395 GST_DEBUG_OBJECT (self, "exclusive mode, can write: %i",
396 self->buffer_frame_count);
397 return self->buffer_frame_count;
400 /* Frames the card hasn't rendered yet */
401 hr = IAudioClient_GetCurrentPadding (self->client, &n_frames_padding);
402 HR_FAILED_RET (hr, IAudioClient::GetCurrentPadding, -1);
404 GST_DEBUG_OBJECT (self, "%i unread frames (padding)", n_frames_padding);
406 /* We can write out these many frames */
407 return self->buffer_frame_count - n_frames_padding;
411 gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
413 GstWasapiSink *self = GST_WASAPI_SINK (asink);
414 gboolean res = FALSE;
415 REFERENCE_TIME latency_rt;
416 guint bpf, rate, devicep_frames;
419 if (self->sharemode == AUDCLNT_SHAREMODE_SHARED &&
420 gst_wasapi_util_have_audioclient3 ()) {
421 if (!gst_wasapi_util_initialize_audioclient3 (GST_ELEMENT (self), spec,
422 (IAudioClient3 *) self->client, self->mix_format, self->low_latency,
426 if (!gst_wasapi_util_initialize_audioclient (GST_ELEMENT (self), spec,
427 self->client, self->mix_format, self->sharemode, self->low_latency,
432 bpf = GST_AUDIO_INFO_BPF (&spec->info);
433 rate = GST_AUDIO_INFO_RATE (&spec->info);
435 /* Total size of the allocated buffer that we will write to */
436 hr = IAudioClient_GetBufferSize (self->client, &self->buffer_frame_count);
437 HR_FAILED_GOTO (hr, IAudioClient::GetBufferSize, beach);
439 GST_INFO_OBJECT (self, "buffer size is %i frames, device period is %i "
440 "frames, bpf is %i bytes, rate is %i Hz", self->buffer_frame_count,
441 devicep_frames, bpf, rate);
443 /* Actual latency-time/buffer-time will be different now */
444 spec->segsize = devicep_frames * bpf;
446 /* We need a minimum of 2 segments to ensure glitch-free playback */
447 spec->segtotal = MAX (self->buffer_frame_count * bpf / spec->segsize, 2);
449 GST_INFO_OBJECT (self, "segsize is %i, segtotal is %i", spec->segsize,
452 /* Get latency for logging */
453 hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
454 HR_FAILED_GOTO (hr, IAudioClient::GetStreamLatency, beach);
456 GST_INFO_OBJECT (self, "wasapi stream latency: %" G_GINT64_FORMAT " (%"
457 G_GINT64_FORMAT "ms)", latency_rt, latency_rt / 10000);
459 /* Set the event handler which will trigger writes */
460 hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
461 HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach);
463 /* Get render sink client and start it up */
464 if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self), self->client,
465 &self->render_client)) {
469 GST_INFO_OBJECT (self, "got render client");
471 /* To avoid start-up glitches, before starting the streaming, we fill the
472 * buffer with silence as recommended by the documentation:
473 * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370879%28v=vs.85%29.aspx */
478 n_frames = gst_wasapi_sink_get_can_frames (self);
480 GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
481 ("should have more than %i frames to write", n_frames));
485 len = n_frames * self->mix_format->nBlockAlign;
487 hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames,
489 HR_FAILED_GOTO (hr, IAudioRenderClient::GetBuffer, beach);
491 GST_DEBUG_OBJECT (self, "pre-wrote %i bytes of silence", len);
493 hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames,
494 AUDCLNT_BUFFERFLAGS_SILENT);
495 HR_FAILED_GOTO (hr, IAudioRenderClient::ReleaseBuffer, beach);
498 hr = IAudioClient_Start (self->client);
499 HR_FAILED_GOTO (hr, IAudioClient::Start, beach);
501 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK
502 (self)->ringbuffer, self->positions);
504 /* Increase the thread priority to reduce glitches */
505 self->thread_priority_handle = gst_wasapi_util_set_thread_characteristics ();
510 /* unprepare() is not called if prepare() fails, but we want it to be, so call
511 * it manually when needed */
513 gst_wasapi_sink_unprepare (asink);
519 gst_wasapi_sink_unprepare (GstAudioSink * asink)
521 GstWasapiSink *self = GST_WASAPI_SINK (asink);
523 if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE &&
524 !gst_wasapi_util_have_audioclient3 ())
527 if (self->thread_priority_handle != NULL) {
528 gst_wasapi_util_revert_thread_characteristics
529 (self->thread_priority_handle);
530 self->thread_priority_handle = NULL;
533 if (self->client != NULL) {
534 IAudioClient_Stop (self->client);
537 if (self->render_client != NULL) {
538 IUnknown_Release (self->render_client);
539 self->render_client = NULL;
546 gst_wasapi_sink_write (GstAudioSink * asink, gpointer data, guint length)
548 GstWasapiSink *self = GST_WASAPI_SINK (asink);
551 guint pending = length;
553 while (pending > 0) {
554 guint can_frames, have_frames, n_frames, write_len;
556 WaitForSingleObject (self->event_handle, INFINITE);
558 /* We have N frames to be written out */
559 have_frames = pending / (self->mix_format->nBlockAlign);
560 /* We have can_frames space in the output buffer */
561 can_frames = gst_wasapi_sink_get_can_frames (self);
562 /* We will write out these many frames, and this much length */
563 n_frames = MIN (can_frames, have_frames);
564 write_len = n_frames * self->mix_format->nBlockAlign;
566 GST_DEBUG_OBJECT (self, "total: %i, have_frames: %i (%i bytes), "
567 "can_frames: %i, will write: %i (%i bytes)", self->buffer_frame_count,
568 have_frames, pending, can_frames, n_frames, write_len);
570 hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames,
572 HR_FAILED_AND (hr, IAudioRenderClient::GetBuffer, length = 0; goto beach);
574 memcpy (dst, data, write_len);
576 hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames,
577 self->mute ? AUDCLNT_BUFFERFLAGS_SILENT : 0);
578 HR_FAILED_AND (hr, IAudioRenderClient::ReleaseBuffer, length = 0;
581 pending -= write_len;
590 gst_wasapi_sink_delay (GstAudioSink * asink)
592 GstWasapiSink *self = GST_WASAPI_SINK (asink);
596 hr = IAudioClient_GetCurrentPadding (self->client, &delay);
597 HR_FAILED_RET (hr, IAudioClient::GetCurrentPadding, 0);
603 gst_wasapi_sink_reset (GstAudioSink * asink)
605 GstWasapiSink *self = GST_WASAPI_SINK (asink);
611 hr = IAudioClient_Stop (self->client);
612 HR_FAILED_RET (hr, IAudioClient::Stop,);
614 hr = IAudioClient_Reset (self->client);
615 HR_FAILED_RET (hr, IAudioClient::Reset,);