gstreamer/sys/wasapi/gstwasapisrc.c

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/*
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* Copyright (C) 2018 Centricular Ltd.
* Author: Nirbheek Chauhan <nirbheek@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-wasapisrc
* @title: wasapisrc
*
* Provides audio capture from the Windows Audio Session API available with
* Vista and newer.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v wasapisrc ! fakesink
* ]| Capture from the default audio device and render to fakesink.
*
* |[
* gst-launch-1.0 -v wasapisrc low-latency=true ! fakesink
* ]| Capture from the default audio device with the minimum possible latency and render to fakesink.
*
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "gstwasapisrc.h"
#include <avrt.h>
GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
#define GST_CAT_DEFAULT gst_wasapi_src_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS));
#define DEFAULT_ROLE GST_WASAPI_DEVICE_ROLE_CONSOLE
#define DEFAULT_LOOPBACK FALSE
#define DEFAULT_EXCLUSIVE FALSE
#define DEFAULT_LOW_LATENCY FALSE
#define DEFAULT_AUDIOCLIENT3 FALSE
/* The clock provided by WASAPI is always off and causes buffers to be late
* very quickly on the sink. Disable pending further investigation. */
#define DEFAULT_PROVIDE_CLOCK FALSE
enum
{
PROP_0,
PROP_ROLE,
PROP_DEVICE,
PROP_LOOPBACK,
PROP_EXCLUSIVE,
PROP_LOW_LATENCY,
PROP_AUDIOCLIENT3
};
static void gst_wasapi_src_dispose (GObject * object);
static void gst_wasapi_src_finalize (GObject * object);
static void gst_wasapi_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wasapi_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
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static gboolean gst_wasapi_src_open (GstAudioSrc * asrc);
static gboolean gst_wasapi_src_close (GstAudioSrc * asrc);
static gboolean gst_wasapi_src_prepare (GstAudioSrc * asrc,
GstAudioRingBufferSpec * spec);
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static gboolean gst_wasapi_src_unprepare (GstAudioSrc * asrc);
static guint gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data,
guint length, GstClockTime * timestamp);
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static guint gst_wasapi_src_delay (GstAudioSrc * asrc);
static void gst_wasapi_src_reset (GstAudioSrc * asrc);
#if DEFAULT_PROVIDE_CLOCK
static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
gpointer user_data);
#endif
#define gst_wasapi_src_parent_class parent_class
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G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_AUDIO_SRC);
static void
gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass);
gobject_class->dispose = gst_wasapi_src_dispose;
gobject_class->finalize = gst_wasapi_src_finalize;
gobject_class->set_property = gst_wasapi_src_set_property;
gobject_class->get_property = gst_wasapi_src_get_property;
g_object_class_install_property (gobject_class,
PROP_ROLE,
g_param_spec_enum ("role", "Role",
"Role of the device: communications, multimedia, etc",
GST_WASAPI_DEVICE_TYPE_ROLE, DEFAULT_ROLE, G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_READY));
g_object_class_install_property (gobject_class,
PROP_DEVICE,
g_param_spec_string ("device", "Device",
"WASAPI playback device as a GUID string",
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_LOOPBACK,
g_param_spec_boolean ("loopback", "Loopback recording",
"Open the sink device for loopback recording",
DEFAULT_LOOPBACK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_EXCLUSIVE,
g_param_spec_boolean ("exclusive", "Exclusive mode",
"Open the device in exclusive mode",
DEFAULT_EXCLUSIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_LOW_LATENCY,
g_param_spec_boolean ("low-latency", "Low latency",
"Optimize all settings for lowest latency. Always safe to enable.",
DEFAULT_LOW_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_AUDIOCLIENT3,
g_param_spec_boolean ("use-audioclient3", "Use the AudioClient3 API",
"Whether to use the Windows 10 AudioClient3 API when available",
DEFAULT_AUDIOCLIENT3, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
"Source/Audio/Hardware",
"Stream audio from an audio capture device through WASAPI",
"Nirbheek Chauhan <nirbheek@centricular.com>, "
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_src_get_caps);
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gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_src_open);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_src_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_wasapi_src_read);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_prepare);
gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_unprepare);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_src_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_src_reset);
GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
0, "Windows audio session API source");
gst_type_mark_as_plugin_api (GST_WASAPI_DEVICE_TYPE_ROLE, 0);
}
static void
gst_wasapi_src_init (GstWasapiSrc * self)
{
#if DEFAULT_PROVIDE_CLOCK
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/* override with a custom clock */
if (GST_AUDIO_BASE_SRC (self)->clock)
gst_object_unref (GST_AUDIO_BASE_SRC (self)->clock);
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GST_AUDIO_BASE_SRC (self)->clock = gst_audio_clock_new ("GstWasapiSrcClock",
gst_wasapi_src_get_time, gst_object_ref (self),
(GDestroyNotify) gst_object_unref);
#endif
self->role = DEFAULT_ROLE;
self->sharemode = AUDCLNT_SHAREMODE_SHARED;
self->loopback = DEFAULT_LOOPBACK;
self->low_latency = DEFAULT_LOW_LATENCY;
self->try_audioclient3 = DEFAULT_AUDIOCLIENT3;
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self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
self->cancellable = CreateEvent (NULL, TRUE, FALSE, NULL);
self->client_needs_restart = FALSE;
self->adapter = gst_adapter_new ();
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/* Extra event handles used for loopback */
self->loopback_event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
self->loopback_cancellable = CreateEvent (NULL, TRUE, FALSE, NULL);
CoInitializeEx (NULL, COINIT_MULTITHREADED);
}
static void
gst_wasapi_src_dispose (GObject * object)
{
GstWasapiSrc *self = GST_WASAPI_SRC (object);
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if (self->event_handle != NULL) {
CloseHandle (self->event_handle);
self->event_handle = NULL;
}
if (self->cancellable != NULL) {
CloseHandle (self->cancellable);
self->cancellable = NULL;
}
if (self->client_clock != NULL) {
IUnknown_Release (self->client_clock);
self->client_clock = NULL;
}
if (self->client != NULL) {
IUnknown_Release (self->client);
self->client = NULL;
}
if (self->capture_client != NULL) {
IUnknown_Release (self->capture_client);
self->capture_client = NULL;
}
if (self->loopback_client != NULL) {
IUnknown_Release (self->loopback_client);
self->loopback_client = NULL;
}
if (self->loopback_event_handle != NULL) {
CloseHandle (self->loopback_event_handle);
self->loopback_event_handle = NULL;
}
if (self->loopback_cancellable != NULL) {
CloseHandle (self->loopback_cancellable);
self->loopback_cancellable = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_wasapi_src_finalize (GObject * object)
{
GstWasapiSrc *self = GST_WASAPI_SRC (object);
CoTaskMemFree (self->mix_format);
self->mix_format = NULL;
CoUninitialize ();
g_clear_pointer (&self->cached_caps, gst_caps_unref);
g_clear_pointer (&self->positions, g_free);
g_clear_pointer (&self->device_strid, g_free);
g_object_unref (self->adapter);
self->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_wasapi_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWasapiSrc *self = GST_WASAPI_SRC (object);
switch (prop_id) {
case PROP_ROLE:
self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value));
break;
case PROP_DEVICE:
{
const gchar *device = g_value_get_string (value);
g_free (self->device_strid);
self->device_strid =
device ? g_utf8_to_utf16 (device, -1, NULL, NULL, NULL) : NULL;
break;
}
case PROP_LOOPBACK:
self->loopback = g_value_get_boolean (value);
break;
case PROP_EXCLUSIVE:
self->sharemode = g_value_get_boolean (value)
? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED;
break;
case PROP_LOW_LATENCY:
self->low_latency = g_value_get_boolean (value);
break;
case PROP_AUDIOCLIENT3:
self->try_audioclient3 = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_wasapi_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWasapiSrc *self = GST_WASAPI_SRC (object);
switch (prop_id) {
case PROP_ROLE:
g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role));
break;
case PROP_DEVICE:
g_value_take_string (value, self->device_strid ?
g_utf16_to_utf8 (self->device_strid, -1, NULL, NULL, NULL) : NULL);
break;
case PROP_LOOPBACK:
g_value_set_boolean (value, self->loopback);
break;
case PROP_EXCLUSIVE:
g_value_set_boolean (value,
self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE);
break;
case PROP_LOW_LATENCY:
g_value_set_boolean (value, self->low_latency);
break;
case PROP_AUDIOCLIENT3:
g_value_set_boolean (value, self->try_audioclient3);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_wasapi_src_can_audioclient3 (GstWasapiSrc * self)
{
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return (self->sharemode == AUDCLNT_SHAREMODE_SHARED &&
self->try_audioclient3 && gst_wasapi_util_have_audioclient3 ());
}
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static GstCaps *
gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstWasapiSrc *self = GST_WASAPI_SRC (bsrc);
WAVEFORMATEX *format = NULL;
GstCaps *caps = NULL;
GST_DEBUG_OBJECT (self, "entering get caps");
if (self->cached_caps) {
caps = gst_caps_ref (self->cached_caps);
} else {
GstCaps *template_caps;
gboolean ret;
template_caps = gst_pad_get_pad_template_caps (bsrc->srcpad);
if (!self->client) {
caps = template_caps;
goto out;
}
ret = gst_wasapi_util_get_device_format (GST_ELEMENT (self),
self->sharemode, self->device, self->client, &format);
if (!ret) {
GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
("failed to detect format"));
gst_caps_unref (template_caps);
return NULL;
}
gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
template_caps, &caps, &self->positions);
if (caps == NULL) {
GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("unknown format"));
gst_caps_unref (template_caps);
return NULL;
}
{
gchar *pos_str = gst_audio_channel_positions_to_string (self->positions,
format->nChannels);
GST_INFO_OBJECT (self, "positions are: %s", pos_str);
g_free (pos_str);
}
self->mix_format = format;
gst_caps_replace (&self->cached_caps, caps);
gst_caps_unref (template_caps);
}
if (filter) {
GstCaps *filtered =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = filtered;
}
out:
GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
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}
static gboolean
gst_wasapi_src_open (GstAudioSrc * asrc)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
gboolean res = FALSE;
IAudioClient *client = NULL;
IMMDevice *device = NULL;
IMMDevice *loopback_device = NULL;
if (self->client)
return TRUE;
/* FIXME: Switching the default device does not switch the stream to it,
* even if the old device was unplugged. We need to handle this somehow.
* For example, perhaps we should automatically switch to the new device if
* the default device is changed and a device isn't explicitly selected. */
if (!gst_wasapi_util_get_device_client (GST_ELEMENT (self),
self->loopback ? eRender : eCapture, self->role, self->device_strid,
&device, &client)) {
if (!self->device_strid)
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("Failed to get default device"));
else
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("Failed to open device %S", self->device_strid));
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goto beach;
}
/* An oddness of wasapi loopback feature is that capture client will not
* provide any audio data if there is no outputting sound.
* To workaround this problem, probably we can add timeout around loop
* in this case but it's glitch prone. So, instead of timeout,
* we will keep pusing silence data to into wasapi client so that make audio
* client report audio data in any case
*/
if (!gst_wasapi_util_get_device_client (GST_ELEMENT (self),
eRender, self->role, self->device_strid,
&loopback_device, &self->loopback_client)) {
if (!self->device_strid)
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("Failed to get default device for loopback"));
else
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("Failed to open device %S", self->device_strid));
goto beach;
/* no need to hold this object */
IUnknown_Release (loopback_device);
}
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self->client = client;
self->device = device;
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res = TRUE;
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beach:
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return res;
}
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static gboolean
gst_wasapi_src_close (GstAudioSrc * asrc)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
if (self->device != NULL) {
IUnknown_Release (self->device);
self->device = NULL;
}
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if (self->client != NULL) {
IUnknown_Release (self->client);
self->client = NULL;
}
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if (self->loopback_client != NULL) {
IUnknown_Release (self->loopback_client);
self->loopback_client = NULL;
}
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return TRUE;
}
static gpointer
gst_wasapi_src_loopback_silence_feeding_thread (GstWasapiSrc * self)
{
HRESULT hr;
UINT32 buffer_frames;
gboolean res G_GNUC_UNUSED = FALSE;
BYTE *data;
DWORD dwWaitResult;
HANDLE event_handle[2];
UINT32 padding;
UINT32 n_frames;
/* NOTE: if this task cause glitch, we need to consider thread priority
* adjusing. See gstaudioutilsprivate.c (e.g., AvSetMmThreadCharacteristics)
* for this context */
GST_INFO_OBJECT (self, "Run loopback silence feeding thread");
event_handle[0] = self->loopback_event_handle;
event_handle[1] = self->loopback_cancellable;
hr = IAudioClient_GetBufferSize (self->loopback_client, &buffer_frames);
HR_FAILED_GOTO (hr, IAudioClient::GetBufferSize, beach);
hr = IAudioClient_SetEventHandle (self->loopback_client,
self->loopback_event_handle);
HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach);
/* To avoid start-up glitches, before starting the streaming, we fill the
* buffer with silence as recommended by the documentation:
* https://msdn.microsoft.com/en-us/library/windows/desktop/dd370879%28v=vs.85%29.aspx */
hr = IAudioRenderClient_GetBuffer (self->loopback_render_client,
buffer_frames, &data);
HR_FAILED_GOTO (hr, IAudioRenderClient::GetBuffer, beach);
hr = IAudioRenderClient_ReleaseBuffer (self->loopback_render_client,
buffer_frames, AUDCLNT_BUFFERFLAGS_SILENT);
HR_FAILED_GOTO (hr, IAudioRenderClient::ReleaseBuffer, beach);
hr = IAudioClient_Start (self->loopback_client);
HR_FAILED_GOTO (hr, IAudioClock::Start, beach);
/* There is an OS bug prior to Windows 10, that is loopback capture client
* will not receive event (in case of event-driven mode).
* A guide for workaround this case is that signal it whenever render client
* writes data.
* See https://docs.microsoft.com/en-us/windows/win32/api/audioclient/nf-audioclient-iaudioclient-initialize
*/
/* Signal for read thread to wakeup */
SetEvent (self->event_handle);
/* Ok, now we are ready for running for feeding silence data */
while (1) {
dwWaitResult = WaitForMultipleObjects (2, event_handle, FALSE, INFINITE);
if (dwWaitResult != WAIT_OBJECT_0 && dwWaitResult != WAIT_OBJECT_0 + 1) {
GST_ERROR_OBJECT (self, "Error waiting for event handle: %x",
(guint) dwWaitResult);
goto stop;
}
/* Stopping was requested from unprepare() */
if (dwWaitResult == WAIT_OBJECT_0 + 1) {
GST_DEBUG_OBJECT (self, "operation was cancelled");
goto stop;
}
hr = IAudioClient_GetCurrentPadding (self->loopback_client, &padding);
HR_FAILED_GOTO (hr, IAudioClock::Start, stop);
if (buffer_frames < padding) {
GST_WARNING_OBJECT (self,
"Current padding %d is too large (buffer size %d)",
padding, buffer_frames);
n_frames = 0;
} else {
n_frames = buffer_frames - padding;
}
hr = IAudioRenderClient_GetBuffer (self->loopback_render_client, n_frames,
&data);
HR_FAILED_GOTO (hr, IAudioRenderClient::GetBuffer, stop);
hr = IAudioRenderClient_ReleaseBuffer (self->loopback_render_client,
n_frames, AUDCLNT_BUFFERFLAGS_SILENT);
HR_FAILED_GOTO (hr, IAudioRenderClient::ReleaseBuffer, stop);
/* Signal for read thread to wakeup */
SetEvent (self->event_handle);
}
stop:
IAudioClient_Stop (self->loopback_client);
beach:
GST_INFO_OBJECT (self, "Terminate loopback silence feeding thread");
return NULL;
}
static gboolean
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gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
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GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
gboolean res = FALSE;
REFERENCE_TIME latency_rt;
guint bpf, rate, devicep_frames, buffer_frames;
HRESULT hr;
CoInitializeEx (NULL, COINIT_MULTITHREADED);
if (gst_wasapi_src_can_audioclient3 (self)) {
if (!gst_wasapi_util_initialize_audioclient3 (GST_ELEMENT (self), spec,
(IAudioClient3 *) self->client, self->mix_format, self->low_latency,
self->loopback, &devicep_frames))
goto beach;
} else {
if (!gst_wasapi_util_initialize_audioclient (GST_ELEMENT (self), spec,
self->client, self->mix_format, self->sharemode, self->low_latency,
self->loopback, &devicep_frames))
goto beach;
}
bpf = GST_AUDIO_INFO_BPF (&spec->info);
rate = GST_AUDIO_INFO_RATE (&spec->info);
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/* Total size in frames of the allocated buffer that we will read from */
hr = IAudioClient_GetBufferSize (self->client, &buffer_frames);
HR_FAILED_GOTO (hr, IAudioClient::GetBufferSize, beach);
GST_INFO_OBJECT (self, "buffer size is %i frames, device period is %i "
"frames, bpf is %i bytes, rate is %i Hz", buffer_frames,
devicep_frames, bpf, rate);
/* Actual latency-time/buffer-time will be different now */
spec->segsize = devicep_frames * bpf;
/* We need a minimum of 2 segments to ensure glitch-free playback */
spec->segtotal = MAX (buffer_frames * bpf / spec->segsize, 2);
GST_INFO_OBJECT (self, "segsize is %i, segtotal is %i", spec->segsize,
spec->segtotal);
/* Get WASAPI latency for logging */
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hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
HR_FAILED_GOTO (hr, IAudioClient::GetStreamLatency, beach);
GST_INFO_OBJECT (self, "wasapi stream latency: %" G_GINT64_FORMAT " (%"
G_GINT64_FORMAT " ms)", latency_rt, latency_rt / 10000);
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/* Set the event handler which will trigger reads */
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hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach);
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/* Get the clock and the clock freq */
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if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client,
&self->client_clock))
goto beach;
hr = IAudioClock_GetFrequency (self->client_clock, &self->client_clock_freq);
HR_FAILED_GOTO (hr, IAudioClock::GetFrequency, beach);
GST_INFO_OBJECT (self, "wasapi clock freq is %" G_GUINT64_FORMAT,
self->client_clock_freq);
/* Get capture source client and start it up */
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if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client,
&self->capture_client)) {
goto beach;
}
/* In case loopback, spawn another dedicated thread for feeding silence data
* into wasapi render client */
if (self->loopback) {
/* don't need to be audioclient3 or low-latency since we will keep pushing
* silence data which is not varying over entire playback */
if (!gst_wasapi_util_initialize_audioclient (GST_ELEMENT (self), spec,
self->loopback_client, self->mix_format, self->sharemode,
FALSE, FALSE, &devicep_frames))
goto beach;
if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self),
self->loopback_client, &self->loopback_render_client)) {
goto beach;
}
self->loopback_thread = g_thread_new ("wasapi-loopback",
(GThreadFunc) gst_wasapi_src_loopback_silence_feeding_thread, self);
}
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hr = IAudioClient_Start (self->client);
HR_FAILED_GOTO (hr, IAudioClock::Start, beach);
self->client_needs_restart = FALSE;
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(self)->ringbuffer, self->positions);
res = TRUE;
/* reset cancellable event handle */
ResetEvent (self->cancellable);
beach:
/* unprepare() is not called if prepare() fails, but we want it to be, so call
* it manually when needed */
if (!res)
gst_wasapi_src_unprepare (asrc);
return res;
}
static gboolean
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gst_wasapi_src_unprepare (GstAudioSrc * asrc)
{
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GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
if (self->client != NULL) {
IAudioClient_Stop (self->client);
}
if (self->capture_client != NULL) {
IUnknown_Release (self->capture_client);
self->capture_client = NULL;
}
if (self->client_clock != NULL) {
IUnknown_Release (self->client_clock);
self->client_clock = NULL;
}
if (self->loopback_thread) {
GST_DEBUG_OBJECT (self, "loopback task thread is stopping");
SetEvent (self->loopback_cancellable);
g_thread_join (self->loopback_thread);
self->loopback_thread = NULL;
ResetEvent (self->loopback_cancellable);
GST_DEBUG_OBJECT (self, "loopback task thread has been stopped");
}
if (self->loopback_render_client != NULL) {
IUnknown_Release (self->loopback_render_client);
self->loopback_render_client = NULL;
}
self->client_clock_freq = 0;
CoUninitialize ();
return TRUE;
}
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static guint
gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{
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GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
HRESULT hr;
gint16 *from = NULL;
guint wanted = length;
guint bpf;
DWORD flags;
GST_OBJECT_LOCK (self);
if (self->client_needs_restart) {
hr = IAudioClient_Start (self->client);
HR_FAILED_ELEMENT_ERROR_AND (hr, IAudioClient::Start, self,
GST_OBJECT_UNLOCK (self); goto err);
self->client_needs_restart = FALSE;
ResetEvent (self->cancellable);
gst_adapter_clear (self->adapter);
}
bpf = self->mix_format->nBlockAlign;
GST_OBJECT_UNLOCK (self);
/* If we've accumulated enough data, return it immediately */
if (gst_adapter_available (self->adapter) >= wanted) {
memcpy (data, gst_adapter_map (self->adapter, wanted), wanted);
gst_adapter_flush (self->adapter, wanted);
GST_DEBUG_OBJECT (self, "Adapter has enough data, returning %i", wanted);
goto out;
}
while (wanted > 0) {
DWORD dwWaitResult;
guint got_frames, avail_frames, n_frames, want_frames, read_len;
HANDLE event_handle[2];
event_handle[0] = self->event_handle;
event_handle[1] = self->cancellable;
/* Wait for data to become available */
dwWaitResult = WaitForMultipleObjects (2, event_handle, FALSE, INFINITE);
if (dwWaitResult != WAIT_OBJECT_0 && dwWaitResult != WAIT_OBJECT_0 + 1) {
GST_ERROR_OBJECT (self, "Error waiting for event handle: %x",
(guint) dwWaitResult);
goto err;
}
/* ::reset was requested */
if (dwWaitResult == WAIT_OBJECT_0 + 1) {
GST_DEBUG_OBJECT (self, "operation was cancelled");
return -1;
}
hr = IAudioCaptureClient_GetBuffer (self->capture_client,
(BYTE **) & from, &got_frames, &flags, NULL, NULL);
if (hr != S_OK) {
if (hr == AUDCLNT_S_BUFFER_EMPTY) {
gchar *msg = gst_wasapi_util_hresult_to_string (hr);
GST_WARNING_OBJECT (self, "IAudioCaptureClient::GetBuffer failed: %s"
", retrying", msg);
g_free (msg);
length = 0;
goto out;
}
HR_FAILED_ELEMENT_ERROR_AND (hr, IAudioCaptureClient::GetBuffer, self,
goto err);
}
if (G_UNLIKELY (flags != 0)) {
/* https://docs.microsoft.com/en-us/windows/win32/api/audioclient/ne-audioclient-_audclnt_bufferflags */
if (flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY)
GST_DEBUG_OBJECT (self, "WASAPI reported discontinuity (glitch?)");
if (flags & AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR)
GST_DEBUG_OBJECT (self, "WASAPI reported a timestamp error");
}
/* Copy all the frames we got into the adapter, and then extract at most
* @wanted size of frames from it. This helps when ::GetBuffer returns more
* data than we can handle right now. */
{
GstBuffer *tmp = gst_buffer_new_allocate (NULL, got_frames * bpf, NULL);
/* If flags has AUDCLNT_BUFFERFLAGS_SILENT, we will ignore the actual
* data and write out silence, see:
* https://docs.microsoft.com/en-us/windows/win32/api/audioclient/ne-audioclient-_audclnt_bufferflags */
if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
memset (from, 0, got_frames * bpf);
gst_buffer_fill (tmp, 0, from, got_frames * bpf);
gst_adapter_push (self->adapter, tmp);
}
/* Release all captured buffers; we copied them above */
hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, got_frames);
from = NULL;
HR_FAILED_ELEMENT_ERROR_AND (hr, IAudioCaptureClient::ReleaseBuffer, self,
goto err);
want_frames = wanted / bpf;
avail_frames = gst_adapter_available (self->adapter) / bpf;
/* Only copy data that will fit into the allocated buffer of size @length */
n_frames = MIN (avail_frames, want_frames);
read_len = n_frames * bpf;
GST_DEBUG_OBJECT (self, "frames captured: %i (%i bytes), "
"can read: %i (%i bytes), will read: %i (%i bytes), "
"adapter has: %i (%i bytes)", got_frames, got_frames * bpf, want_frames,
wanted, n_frames, read_len, avail_frames, avail_frames * bpf);
memcpy (data, gst_adapter_map (self->adapter, read_len), read_len);
gst_adapter_flush (self->adapter, read_len);
wanted -= read_len;
}
out:
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return length;
err:
length = -1;
goto out;
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}
static guint
gst_wasapi_src_delay (GstAudioSrc * asrc)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
guint delay = 0;
HRESULT hr;
hr = IAudioClient_GetCurrentPadding (self->client, &delay);
HR_FAILED_RET (hr, IAudioClock::GetCurrentPadding, 0);
return delay;
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}
static void
gst_wasapi_src_reset (GstAudioSrc * asrc)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
HRESULT hr;
if (!self->client)
return;
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SetEvent (self->cancellable);
GST_OBJECT_LOCK (self);
hr = IAudioClient_Stop (self->client);
HR_FAILED_AND (hr, IAudioClock::Stop, goto err);
hr = IAudioClient_Reset (self->client);
HR_FAILED_AND (hr, IAudioClock::Reset, goto err);
err:
self->client_needs_restart = TRUE;
GST_OBJECT_UNLOCK (self);
}
#if DEFAULT_PROVIDE_CLOCK
static GstClockTime
gst_wasapi_src_get_time (GstClock * clock, gpointer user_data)
{
GstWasapiSrc *self = GST_WASAPI_SRC (user_data);
HRESULT hr;
guint64 devpos;
GstClockTime result;
if (G_UNLIKELY (self->client_clock == NULL))
return GST_CLOCK_TIME_NONE;
hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL);
HR_FAILED_RET (hr, IAudioClock::GetPosition, GST_CLOCK_TIME_NONE);
result = gst_util_uint64_scale_int (devpos, GST_SECOND,
self->client_clock_freq);
/*
GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT
" frequency = %" G_GUINT64_FORMAT
" result = %" G_GUINT64_FORMAT " ms",
devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result));
*/
return result;
}
#endif