2 * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
3 * Copyright (C) 2018 Centricular Ltd.
4 * Author: Nirbheek Chauhan <nirbheek@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * SECTION:element-wasapisrc
26 * Provides audio capture from the Windows Audio Session API available with
29 * ## Example pipelines
31 * gst-launch-1.0 -v wasapisrc ! fakesink
32 * ]| Capture from the default audio device and render to fakesink.
39 #include "gstwasapisrc.h"
43 GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
44 #define GST_CAT_DEFAULT gst_wasapi_src_debug
46 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
49 GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS));
51 #define DEFAULT_ROLE GST_WASAPI_DEVICE_ROLE_CONSOLE
52 #define DEFAULT_EXCLUSIVE FALSE
53 #define DEFAULT_LOW_LATENCY FALSE
54 #define DEFAULT_AUDIOCLIENT3 FALSE
66 static void gst_wasapi_src_dispose (GObject * object);
67 static void gst_wasapi_src_finalize (GObject * object);
68 static void gst_wasapi_src_set_property (GObject * object, guint prop_id,
69 const GValue * value, GParamSpec * pspec);
70 static void gst_wasapi_src_get_property (GObject * object, guint prop_id,
71 GValue * value, GParamSpec * pspec);
73 static GstCaps *gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
75 static gboolean gst_wasapi_src_open (GstAudioSrc * asrc);
76 static gboolean gst_wasapi_src_close (GstAudioSrc * asrc);
77 static gboolean gst_wasapi_src_prepare (GstAudioSrc * asrc,
78 GstAudioRingBufferSpec * spec);
79 static gboolean gst_wasapi_src_unprepare (GstAudioSrc * asrc);
80 static guint gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data,
81 guint length, GstClockTime * timestamp);
82 static guint gst_wasapi_src_delay (GstAudioSrc * asrc);
83 static void gst_wasapi_src_reset (GstAudioSrc * asrc);
85 static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
88 #define gst_wasapi_src_parent_class parent_class
89 G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_AUDIO_SRC);
92 gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
94 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
95 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
96 GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
97 GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass);
99 gobject_class->dispose = gst_wasapi_src_dispose;
100 gobject_class->finalize = gst_wasapi_src_finalize;
101 gobject_class->set_property = gst_wasapi_src_set_property;
102 gobject_class->get_property = gst_wasapi_src_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_string ("device", "Device",
114 "WASAPI playback device as a GUID string",
115 NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
117 g_object_class_install_property (gobject_class,
119 g_param_spec_boolean ("exclusive", "Exclusive mode",
120 "Open the device in exclusive mode",
121 DEFAULT_EXCLUSIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
123 g_object_class_install_property (gobject_class,
125 g_param_spec_boolean ("low-latency", "Low latency",
126 "Optimize all settings for lowest latency",
127 DEFAULT_LOW_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
129 g_object_class_install_property (gobject_class,
131 g_param_spec_boolean ("use-audioclient3", "Use the AudioClient3 API",
132 "Whether to use the Windows 10 AudioClient3 API when available",
133 DEFAULT_AUDIOCLIENT3, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
135 gst_element_class_add_static_pad_template (gstelement_class, &src_template);
136 gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
138 "Stream audio from an audio capture device through WASAPI",
139 "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
141 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_src_get_caps);
143 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_src_open);
144 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_src_close);
145 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_wasapi_src_read);
146 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_prepare);
147 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_unprepare);
148 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_src_delay);
149 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_src_reset);
151 GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
152 0, "Windows audio session API source");
156 gst_wasapi_src_init (GstWasapiSrc * self)
158 /* override with a custom clock */
159 if (GST_AUDIO_BASE_SRC (self)->clock)
160 gst_object_unref (GST_AUDIO_BASE_SRC (self)->clock);
162 GST_AUDIO_BASE_SRC (self)->clock = gst_audio_clock_new ("GstWasapiSrcClock",
163 gst_wasapi_src_get_time, gst_object_ref (self),
164 (GDestroyNotify) gst_object_unref);
166 self->role = DEFAULT_ROLE;
167 self->sharemode = AUDCLNT_SHAREMODE_SHARED;
168 self->low_latency = DEFAULT_LOW_LATENCY;
169 self->try_audioclient3 = DEFAULT_AUDIOCLIENT3;
170 self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
176 gst_wasapi_src_dispose (GObject * object)
178 GstWasapiSrc *self = GST_WASAPI_SRC (object);
180 if (self->event_handle != NULL) {
181 CloseHandle (self->event_handle);
182 self->event_handle = NULL;
185 if (self->client_clock != NULL) {
186 IUnknown_Release (self->client_clock);
187 self->client_clock = NULL;
190 if (self->client != NULL) {
191 IUnknown_Release (self->client);
195 if (self->capture_client != NULL) {
196 IUnknown_Release (self->capture_client);
197 self->capture_client = NULL;
200 G_OBJECT_CLASS (parent_class)->dispose (object);
204 gst_wasapi_src_finalize (GObject * object)
206 GstWasapiSrc *self = GST_WASAPI_SRC (object);
208 g_clear_pointer (&self->mix_format, CoTaskMemFree);
212 g_clear_pointer (&self->cached_caps, gst_caps_unref);
213 g_clear_pointer (&self->positions, g_free);
214 g_clear_pointer (&self->device_strid, g_free);
216 G_OBJECT_CLASS (parent_class)->finalize (object);
220 gst_wasapi_src_set_property (GObject * object, guint prop_id,
221 const GValue * value, GParamSpec * pspec)
223 GstWasapiSrc *self = GST_WASAPI_SRC (object);
227 self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value));
231 const gchar *device = g_value_get_string (value);
232 g_free (self->device_strid);
234 device ? g_utf8_to_utf16 (device, -1, NULL, NULL, NULL) : NULL;
238 self->sharemode = g_value_get_boolean (value)
239 ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED;
241 case PROP_LOW_LATENCY:
242 self->low_latency = g_value_get_boolean (value);
244 case PROP_AUDIOCLIENT3:
245 self->try_audioclient3 = g_value_get_boolean (value);
248 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
254 gst_wasapi_src_get_property (GObject * object, guint prop_id,
255 GValue * value, GParamSpec * pspec)
257 GstWasapiSrc *self = GST_WASAPI_SRC (object);
261 g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role));
264 g_value_take_string (value, self->device_strid ?
265 g_utf16_to_utf8 (self->device_strid, -1, NULL, NULL, NULL) : NULL);
268 g_value_set_boolean (value,
269 self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE);
271 case PROP_LOW_LATENCY:
272 g_value_set_boolean (value, self->low_latency);
274 case PROP_AUDIOCLIENT3:
275 g_value_set_boolean (value, self->try_audioclient3);
278 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
284 gst_wasapi_src_can_audioclient3 (GstWasapiSrc * self)
286 if (self->sharemode == AUDCLNT_SHAREMODE_SHARED &&
287 self->try_audioclient3 && gst_wasapi_util_have_audioclient3 ())
293 gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
295 GstWasapiSrc *self = GST_WASAPI_SRC (bsrc);
296 WAVEFORMATEX *format = NULL;
297 GstCaps *caps = NULL;
299 GST_DEBUG_OBJECT (self, "entering get caps");
301 if (self->cached_caps) {
302 caps = gst_caps_ref (self->cached_caps);
304 GstCaps *template_caps;
307 template_caps = gst_pad_get_pad_template_caps (bsrc->srcpad);
310 gst_wasapi_src_open (GST_AUDIO_SRC (bsrc));
312 ret = gst_wasapi_util_get_device_format (GST_ELEMENT (self),
313 self->sharemode, self->device, self->client, &format);
315 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
316 ("failed to detect format"));
320 gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
321 template_caps, &caps, &self->positions);
323 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("unknown format"));
328 gchar *pos_str = gst_audio_channel_positions_to_string (self->positions,
330 GST_INFO_OBJECT (self, "positions are: %s", pos_str);
334 self->mix_format = format;
335 gst_caps_replace (&self->cached_caps, caps);
336 gst_caps_unref (template_caps);
341 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
342 gst_caps_unref (caps);
346 GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps);
353 gst_wasapi_src_open (GstAudioSrc * asrc)
355 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
356 gboolean res = FALSE;
357 IAudioClient *client = NULL;
358 IMMDevice *device = NULL;
363 /* FIXME: Switching the default device does not switch the stream to it,
364 * even if the old device was unplugged. We need to handle this somehow.
365 * For example, perhaps we should automatically switch to the new device if
366 * the default device is changed and a device isn't explicitly selected. */
367 if (!gst_wasapi_util_get_device_client (GST_ELEMENT (self), TRUE,
368 self->role, self->device_strid, &device, &client)) {
369 if (!self->device_strid)
370 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
371 ("Failed to get default device"));
373 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
374 ("Failed to open device %S", self->device_strid));
378 self->client = client;
379 self->device = device;
388 gst_wasapi_src_close (GstAudioSrc * asrc)
390 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
392 if (self->device != NULL) {
393 IUnknown_Release (self->device);
397 if (self->client != NULL) {
398 IUnknown_Release (self->client);
406 gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
408 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
409 gboolean res = FALSE;
410 REFERENCE_TIME latency_rt;
411 guint bpf, rate, devicep_frames, buffer_frames;
414 if (gst_wasapi_src_can_audioclient3 (self)) {
415 if (!gst_wasapi_util_initialize_audioclient3 (GST_ELEMENT (self), spec,
416 (IAudioClient3 *) self->client, self->mix_format, self->low_latency,
420 if (!gst_wasapi_util_initialize_audioclient (GST_ELEMENT (self), spec,
421 self->client, self->mix_format, self->sharemode, self->low_latency,
426 bpf = GST_AUDIO_INFO_BPF (&spec->info);
427 rate = GST_AUDIO_INFO_RATE (&spec->info);
429 /* Total size in frames of the allocated buffer that we will read from */
430 hr = IAudioClient_GetBufferSize (self->client, &buffer_frames);
431 HR_FAILED_GOTO (hr, IAudioClient::GetBufferSize, beach);
433 GST_INFO_OBJECT (self, "buffer size is %i frames, device period is %i "
434 "frames, bpf is %i bytes, rate is %i Hz", buffer_frames,
435 devicep_frames, bpf, rate);
437 /* Actual latency-time/buffer-time will be different now */
438 spec->segsize = devicep_frames * bpf;
440 /* We need a minimum of 2 segments to ensure glitch-free playback */
441 spec->segtotal = MAX (self->buffer_frame_count * bpf / spec->segsize, 2);
443 GST_INFO_OBJECT (self, "segsize is %i, segtotal is %i", spec->segsize,
446 /* Get WASAPI latency for logging */
447 hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
448 HR_FAILED_GOTO (hr, IAudioClient::GetStreamLatency, beach);
450 GST_INFO_OBJECT (self, "wasapi stream latency: %" G_GINT64_FORMAT " (%"
451 G_GINT64_FORMAT " ms)", latency_rt, latency_rt / 10000);
453 /* Set the event handler which will trigger reads */
454 hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
455 HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach);
457 /* Get the clock and the clock freq */
458 if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client,
459 &self->client_clock))
462 hr = IAudioClock_GetFrequency (self->client_clock, &self->client_clock_freq);
463 HR_FAILED_GOTO (hr, IAudioClock::GetFrequency, beach);
465 /* Get capture source client and start it up */
466 if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client,
467 &self->capture_client)) {
471 hr = IAudioClient_Start (self->client);
472 HR_FAILED_GOTO (hr, IAudioClock::Start, beach);
474 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
475 (self)->ringbuffer, self->positions);
477 /* Increase the thread priority to reduce glitches */
478 self->thread_priority_handle = gst_wasapi_util_set_thread_characteristics ();
482 /* unprepare() is not called if prepare() fails, but we want it to be, so call
483 * it manually when needed */
485 gst_wasapi_src_unprepare (asrc);
491 gst_wasapi_src_unprepare (GstAudioSrc * asrc)
493 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
495 if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE &&
496 !gst_wasapi_src_can_audioclient3 (self))
499 if (self->thread_priority_handle != NULL) {
500 gst_wasapi_util_revert_thread_characteristics
501 (self->thread_priority_handle);
502 self->thread_priority_handle = NULL;
505 if (self->client != NULL) {
506 IAudioClient_Stop (self->client);
509 if (self->capture_client != NULL) {
510 IUnknown_Release (self->capture_client);
511 self->capture_client = NULL;
514 if (self->client_clock != NULL) {
515 IUnknown_Release (self->client_clock);
516 self->client_clock = NULL;
519 self->client_clock_freq = 0;
525 gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length,
526 GstClockTime * timestamp)
528 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
531 guint wanted = length;
535 guint have_frames, n_frames, want_frames, read_len;
537 /* Wait for data to become available */
538 WaitForSingleObject (self->event_handle, INFINITE);
540 hr = IAudioCaptureClient_GetBuffer (self->capture_client,
541 (BYTE **) & from, &have_frames, &flags, NULL, NULL);
543 gchar *msg = gst_wasapi_util_hresult_to_string (hr);
544 if (hr == AUDCLNT_S_BUFFER_EMPTY)
545 GST_WARNING_OBJECT (self, "IAudioCaptureClient::GetBuffer failed: %s"
548 GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer failed: %s",
556 GST_INFO_OBJECT (self, "buffer flags=%#08x", (guint) flags);
558 /* XXX: How do we handle AUDCLNT_BUFFERFLAGS_SILENT? We're supposed to write
559 * out silence when that flag is set? See:
560 * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370800(v=vs.85).aspx */
562 if (flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY)
563 GST_WARNING_OBJECT (self, "WASAPI reported glitch in buffer");
565 want_frames = wanted / self->mix_format->nBlockAlign;
567 /* If GetBuffer is returning more frames than we can handle, all we can do is
568 * hope that this is temporary and that things will settle down later. */
569 if (G_UNLIKELY (have_frames > want_frames))
570 GST_WARNING_OBJECT (self, "captured too many frames: have %i, want %i",
571 have_frames, want_frames);
573 /* Only copy data that will fit into the allocated buffer of size @length */
574 n_frames = MIN (have_frames, want_frames);
575 read_len = n_frames * self->mix_format->nBlockAlign;
578 guint bpf = self->mix_format->nBlockAlign;
579 GST_DEBUG_OBJECT (self, "have: %i (%i bytes), can read: %i (%i bytes), "
580 "will read: %i (%i bytes)", have_frames, have_frames * bpf,
581 want_frames, wanted, n_frames, read_len);
584 memcpy (data, from, read_len);
587 /* Always release all captured buffers if we've captured any at all */
588 hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, have_frames);
589 HR_FAILED_AND (hr, IAudioClock::ReleaseBuffer, goto beach);
599 gst_wasapi_src_delay (GstAudioSrc * asrc)
601 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
605 hr = IAudioClient_GetCurrentPadding (self->client, &delay);
606 HR_FAILED_RET (hr, IAudioClock::GetCurrentPadding, 0);
612 gst_wasapi_src_reset (GstAudioSrc * asrc)
614 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
620 hr = IAudioClient_Stop (self->client);
621 HR_FAILED_RET (hr, IAudioClock::Stop,);
623 hr = IAudioClient_Reset (self->client);
624 HR_FAILED_RET (hr, IAudioClock::Reset,);
628 gst_wasapi_src_get_time (GstClock * clock, gpointer user_data)
630 GstWasapiSrc *self = GST_WASAPI_SRC (user_data);
635 if (G_UNLIKELY (self->client_clock == NULL))
636 return GST_CLOCK_TIME_NONE;
638 hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL);
639 HR_FAILED_RET (hr, IAudioClock::GetPosition, GST_CLOCK_TIME_NONE);
641 result = gst_util_uint64_scale_int (devpos, GST_SECOND,
642 self->client_clock_freq);
645 GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT
646 " frequency = %" G_GUINT64_FORMAT
647 " result = %" G_GUINT64_FORMAT " ms",
648 devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result));