gstreamer/sys/wasapi/gstwasapisink.c
Nirbheek Chauhan 1450851095 wasapi: Rewrite most of the code to make it work
Both the source and the sink elements were broken in a number of ways:

* prepare() was assuming that the format was always S16LE 2ch 44.1KHz.
  We now probe the preferred format with GetMixFormat().
* Device initialization was done with the wrong buffer size
  (buffer_time is in microseconds, not nanoseconds).
* sink_write() and src_read() were just plain wrong and would never
  write or read anything useful.
* Some functions in prepare() were always returning FALSE which meant
  trying to use the elements would *always* fail.
* get_caps() and delay() were not implemented at all.

TODO: support for >2 channels
TODO: pro-audio low-latency
TODO: SPDIF and other encoded passthroughs

Three new properties are now implemented: role, mute, and device.

* 'role' designates the stream role of the initialized device, see:
   https://msdn.microsoft.com/en-us/library/windows/desktop/dd370842(v=vs.85).aspx
* 'device' is a system-wide GUIDesque string for a specific device.
* 'mute' is a sink property and simply mutes it.

On my Windows 8.1 system, the lowest latency that works is:

  wasapisrc buffer-time=20000
  wasapisink buffer-time=10000

aka, 20ms and 10ms respectively. These values are close to the lowest
possible with the IAudioClient interface. Further improvements require
porting to IAudioClient2 or IAudioClient3.

https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/low-latency-audio
2018-01-22 14:18:53 +05:30

510 lines
15 KiB
C

/*
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* Copyright (C) 2013 Collabora Ltd.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
* 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-wasapisink
* @title: wasapisink
*
* Provides audio playback using the Windows Audio Session API available with
* Vista and newer.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink
* ]| Generate 20 ms buffers and render to the default audio device.
*
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "gstwasapisink.h"
#include <mmdeviceapi.h>
GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
#define GST_CAT_DEFAULT gst_wasapi_sink_debug
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_FORMATS_ALL ", "
"layout = (string) interleaved, "
"rate = " GST_AUDIO_RATE_RANGE ", channels = (int) [1, 2]"));
#define DEFAULT_ROLE GST_WASAPI_DEVICE_ROLE_CONSOLE
#define DEFAULT_MUTE FALSE
enum
{
PROP_0,
PROP_ROLE,
PROP_MUTE,
PROP_DEVICE
};
static void gst_wasapi_sink_dispose (GObject * object);
static void gst_wasapi_sink_finalize (GObject * object);
static void gst_wasapi_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wasapi_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_wasapi_sink_get_caps (GstBaseSink * bsink,
GstCaps * filter);
static gboolean gst_wasapi_sink_prepare (GstAudioSink * asink,
GstAudioRingBufferSpec * spec);
static gboolean gst_wasapi_sink_unprepare (GstAudioSink * asink);
static gboolean gst_wasapi_sink_open (GstAudioSink * asink);
static gboolean gst_wasapi_sink_close (GstAudioSink * asink);
static gint gst_wasapi_sink_write (GstAudioSink * asink,
gpointer data, guint length);
static guint gst_wasapi_sink_delay (GstAudioSink * asink);
static void gst_wasapi_sink_reset (GstAudioSink * asink);
#define gst_wasapi_sink_parent_class parent_class
G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_SINK);
static void
gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass);
gobject_class->dispose = gst_wasapi_sink_dispose;
gobject_class->finalize = gst_wasapi_sink_finalize;
gobject_class->set_property = gst_wasapi_sink_set_property;
gobject_class->get_property = gst_wasapi_sink_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_MUTE,
g_param_spec_boolean ("mute", "Mute", "Mute state of this stream",
DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_PLAYING));
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));
gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
"Sink/Audio",
"Stream audio to an audio capture device through WASAPI",
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_sink_get_caps);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_prepare);
gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_unprepare);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_sink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_sink_close);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_wasapi_sink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_sink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_sink_reset);
GST_DEBUG_CATEGORY_INIT (gst_wasapi_sink_debug, "wasapisink",
0, "Windows audio session API sink");
}
static void
gst_wasapi_sink_init (GstWasapiSink * self)
{
self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
CoInitialize (NULL);
}
static void
gst_wasapi_sink_dispose (GObject * object)
{
GstWasapiSink *self = GST_WASAPI_SINK (object);
if (self->event_handle != NULL) {
CloseHandle (self->event_handle);
self->event_handle = NULL;
}
G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->dispose (object);
}
static void
gst_wasapi_sink_finalize (GObject * object)
{
CoUninitialize ();
G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object);
}
static void
gst_wasapi_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWasapiSink *self = GST_WASAPI_SINK (object);
switch (prop_id) {
case PROP_ROLE:
self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value));
break;
case PROP_MUTE:
self->mute = g_value_get_boolean (value);
break;
case PROP_DEVICE:
{
gchar *device = g_value_get_string (value);
g_free (self->device);
self->device =
device ? g_utf8_to_utf16 (device, 0, NULL, NULL, NULL) : NULL;
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_wasapi_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWasapiSink *self = GST_WASAPI_SINK (object);
switch (prop_id) {
case PROP_ROLE:
g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role));
break;
case PROP_MUTE:
g_value_set_boolean (value, self->mute);
break;
case PROP_DEVICE:
g_value_take_string (value, self->device ?
g_utf16_to_utf8 (self->device, 0, NULL, NULL, NULL) : NULL);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_wasapi_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
{
GstWasapiSink *self = GST_WASAPI_SINK (bsink);
WAVEFORMATEX *format = NULL;
GstCaps *caps = NULL;
HRESULT hr;
GST_DEBUG_OBJECT (self, "entering get caps");
if (self->cached_caps) {
caps = gst_caps_ref (self->cached_caps);
} else {
GstCaps *template_caps;
template_caps = gst_pad_get_pad_template_caps (bsink->sinkpad);
if (!self->client)
gst_wasapi_sink_open (GST_AUDIO_SINK (bsink));
hr = IAudioClient_GetMixFormat (self->client, &format);
if (hr != S_OK || format == NULL) {
GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
("GetMixFormat failed: %s", gst_wasapi_util_hresult_to_string (hr)));
goto out;
}
caps =
gst_wasapi_util_waveformatex_to_caps ((WAVEFORMATEXTENSIBLE *) format,
template_caps);
if (caps == NULL) {
GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("unknown format"));
goto out;
}
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;
}
GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps);
out:
return caps;
}
static gboolean
gst_wasapi_sink_open (GstAudioSink * asink)
{
GstWasapiSink *self = GST_WASAPI_SINK (asink);
gboolean res = FALSE;
IAudioClient *client = NULL;
GST_DEBUG_OBJECT (self, "opening device");
if (self->client)
return TRUE;
if (!gst_wasapi_util_get_device_client (GST_ELEMENT (self), FALSE,
self->role, self->device, &client)) {
if (!self->device)
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));
goto beach;
}
self->client = client;
res = TRUE;
beach:
return res;
}
static gboolean
gst_wasapi_sink_close (GstAudioSink * asink)
{
GstWasapiSink *self = GST_WASAPI_SINK (asink);
if (self->client != NULL) {
IUnknown_Release (self->client);
self->client = NULL;
}
return TRUE;
}
static gboolean
gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
{
GstWasapiSink *self = GST_WASAPI_SINK (asink);
gboolean res = FALSE;
HRESULT hr;
REFERENCE_TIME latency_rt;
IAudioRenderClient *render_client = NULL;
hr = IAudioClient_Initialize (self->client, AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
spec->buffer_time * 10, 0, self->mix_format, NULL);
if (hr != S_OK) {
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("IAudioClient::Initialize () failed: %s",
gst_wasapi_util_hresult_to_string (hr)));
goto beach;
}
/* Get latency for logging */
hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetStreamLatency failed");
goto beach;
}
GST_INFO_OBJECT (self, "wasapi stream latency: %" G_GINT64_FORMAT " (%"
G_GINT64_FORMAT "ms)", latency_rt, latency_rt / 10000);
/* Set the event handler which will trigger writes */
hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle failed");
goto beach;
}
/* Total size of the allocated buffer that we will write to
* XXX: Will this ever change while playing? */
hr = IAudioClient_GetBufferSize (self->client, &self->buffer_frame_count);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetBufferSize failed");
goto beach;
}
GST_INFO_OBJECT (self, "frame count is %i, blockAlign is %i, "
"buffer_time is %" G_GINT64_FORMAT, self->buffer_frame_count,
self->mix_format->nBlockAlign, spec->buffer_time);
/* Get render sink client and start it up */
if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self), self->client,
&render_client)) {
goto beach;
}
GST_INFO_OBJECT (self, "got render client");
hr = IAudioClient_Start (self->client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
goto beach;
}
self->render_client = render_client;
render_client = NULL;
res = TRUE;
beach:
if (render_client != NULL)
IUnknown_Release (render_client);
return res;
}
static gboolean
gst_wasapi_sink_unprepare (GstAudioSink * asink)
{
GstWasapiSink *self = GST_WASAPI_SINK (asink);
if (self->client != NULL) {
IAudioClient_Stop (self->client);
}
if (self->render_client != NULL) {
IUnknown_Release (self->render_client);
self->render_client = NULL;
}
return TRUE;
}
static gint
gst_wasapi_sink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstWasapiSink *self = GST_WASAPI_SINK (asink);
HRESULT hr;
gint16 *dst = NULL;
guint pending = length;
while (pending > 0) {
guint have_frames, can_frames, n_frames, n_frames_padding, write_len;
/* We have N frames to be written out */
have_frames = pending / (self->mix_format->nBlockAlign);
WaitForSingleObject (self->event_handle, INFINITE);
/* Frames the card hasn't rendered yet */
hr = IAudioClient_GetCurrentPadding (self->client, &n_frames_padding);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetCurrentPadding failed: %s",
gst_wasapi_util_hresult_to_string (hr));
length = 0;
goto beach;
}
/* We can write out these many frames */
can_frames = self->buffer_frame_count - n_frames_padding;
/* We will write out these many frames, and this much length */
n_frames = MIN (can_frames, have_frames);
write_len = n_frames * self->mix_format->nBlockAlign;
GST_TRACE_OBJECT (self, "total: %i, unread: %i, have: %i (%i bytes), "
"will write: %i (%i bytes)", self->buffer_frame_count, n_frames_padding,
have_frames, pending, n_frames, write_len);
hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames,
(BYTE **) & dst);
if (hr != S_OK) {
GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
("IAudioRenderClient::GetBuffer failed: %s",
gst_wasapi_util_hresult_to_string (hr)));
length = 0;
goto beach;
}
memcpy (dst, data, write_len);
hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames,
self->mute ? AUDCLNT_BUFFERFLAGS_SILENT : 0);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioRenderClient::ReleaseBuffer failed: %s",
gst_wasapi_util_hresult_to_string (hr));
length = 0;
goto beach;
}
pending -= write_len;
}
beach:
return length;
}
static guint
gst_wasapi_sink_delay (GstAudioSink * asink)
{
GstWasapiSink *self = GST_WASAPI_SINK (asink);
guint delay = 0;
HRESULT hr;
hr = IAudioClient_GetCurrentPadding (self->client, &delay);
if (hr != S_OK) {
GST_ELEMENT_ERROR (self, RESOURCE, READ, (NULL),
("IAudioClient::GetCurrentPadding failed %s",
gst_wasapi_util_hresult_to_string (hr)));
}
return delay;
}
static void
gst_wasapi_sink_reset (GstAudioSink * asink)
{
GstWasapiSink *self = GST_WASAPI_SINK (asink);
HRESULT hr;
if (self->client) {
hr = IAudioClient_Stop (self->client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Stop () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
return;
}
hr = IAudioClient_Reset (self->client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Reset () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
return;
}
}
}