gstreamer/sys/wasapi/gstwasapisrc.c
2013-04-23 18:57:04 +02:00

405 lines
11 KiB
C

/*
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.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
*
* Provides audio capture from the Windows Audio Session API available with
* Vista and newer.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 -v wasapisrc ! fakesink
* ]| Capture from the default audio device and render to fakesink.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "gstwasapisrc.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 ("audio/x-raw, "
"format = (string) S16LE, "
"layout = (string) interleaved, "
"rate = (int) 44100, " "channels = (int) 1"));
static void gst_wasapi_src_dispose (GObject * object);
static void gst_wasapi_src_finalize (GObject * object);
static GstCaps * gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
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);
static gboolean gst_wasapi_src_unprepare (GstAudioSrc * asrc);
static guint gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length, GstClockTime * timestamp);
static guint gst_wasapi_src_delay (GstAudioSrc * asrc);
static void gst_wasapi_src_reset (GstAudioSrc * asrc);
static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
gpointer user_data);
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);
GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass);
gobject_class->dispose = gst_wasapi_src_dispose;
gobject_class->finalize = gst_wasapi_src_finalize;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
"Source/Audio",
"Stream audio from an audio capture device through WASAPI",
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_src_get_caps);
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);
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");
}
static void
gst_wasapi_src_init (GstWasapiSrc * self)
{
/* override with a custom clock */
if (GST_AUDIO_BASE_SRC (self)->clock)
gst_object_unref (GST_AUDIO_BASE_SRC (self)->clock);
GST_AUDIO_BASE_SRC (self)->clock = gst_audio_clock_new ("GstWasapiSrcClock",
gst_wasapi_src_get_time, gst_object_ref (self),
(GDestroyNotify) gst_object_unref);
self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
CoInitialize (NULL);
}
static void
gst_wasapi_src_dispose (GObject * object)
{
GstWasapiSrc *self = GST_WASAPI_SRC (object);
if (self->event_handle != NULL) {
CloseHandle (self->event_handle);
self->event_handle = NULL;
}
G_OBJECT_CLASS (gst_wasapi_src_parent_class)->dispose (object);
}
static void
gst_wasapi_src_finalize (GObject * object)
{
CoUninitialize ();
G_OBJECT_CLASS (gst_wasapi_src_parent_class)->finalize (object);
}
static GstCaps *
gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
{
/* TODO: Implement */
return NULL;
}
static gboolean
gst_wasapi_src_open (GstAudioSrc * asrc)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
gboolean res = FALSE;
IAudioClient * client = NULL;
if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self), TRUE, &client)) {
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("Failed to get default device"));
goto beach;
}
self->client = client;
res = TRUE;
beach:
return res;
}
static gboolean
gst_wasapi_src_close (GstAudioSrc * asrc)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
if (self->client != NULL) {
IUnknown_Release (self->client);
self->client = NULL;
}
return TRUE;
}
static gboolean
gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
gboolean res = FALSE;
IAudioClock *client_clock = NULL;
guint64 client_clock_freq = 0;
IAudioCaptureClient *capture_client = NULL;
REFERENCE_TIME latency_rt, def_period, min_period;
WAVEFORMATEXTENSIBLE format;
HRESULT hr;
hr = IAudioClient_GetDevicePeriod (self->client, &def_period, &min_period);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetDevicePeriod () failed");
goto beach;
}
gst_wasapi_util_audio_info_to_waveformatex (&spec->info, &format);
self->info = spec->info;
hr = IAudioClient_Initialize (self->client, AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
spec->buffer_time / 100, 0, (WAVEFORMATEX *) & 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;
}
hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetStreamLatency () failed");
goto beach;
}
GST_INFO_OBJECT (self, "default period: %d (%d ms), "
"minimum period: %d (%d ms), "
"latency: %d (%d ms)",
(guint32) def_period, (guint32) def_period / 10000,
(guint32) min_period, (guint32) min_period / 10000,
(guint32) latency_rt, (guint32) latency_rt / 10000);
/* FIXME: What to do with the latency? */
hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
goto beach;
}
if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client,
&client_clock)) {
goto beach;
}
hr = IAudioClock_GetFrequency (client_clock, &client_clock_freq);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClock::GetFrequency () failed");
goto beach;
}
if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client,
&capture_client)) {
goto beach;
}
hr = IAudioClient_Start (self->client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
goto beach;
}
self->client_clock = client_clock;
self->client_clock_freq = client_clock_freq;
self->capture_client = capture_client;
res = TRUE;
beach:
if (!res) {
if (capture_client != NULL)
IUnknown_Release (capture_client);
if (client_clock != NULL)
IUnknown_Release (client_clock);
}
return res;
}
static gboolean
gst_wasapi_src_unprepare (GstAudioSrc * asrc)
{
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;
}
return TRUE;
}
static guint
gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
HRESULT hr;
gint16 *samples = NULL;
guint32 nsamples = 0, length_samples;
DWORD flags = 0;
guint64 devpos;
guint i;
gint16 *dst;
WaitForSingleObject (self->event_handle, INFINITE);
do {
hr = IAudioCaptureClient_GetBuffer (self->capture_client,
(BYTE **) & samples, &nsamples, &flags, &devpos, NULL);
}
while (hr == AUDCLNT_S_BUFFER_EMPTY);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
length = 0;
goto beach;
}
if (flags != 0) {
GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
devpos, (guint) flags);
}
length_samples = length / self->info.bpf;
nsamples = MIN (length_samples, nsamples);
length = nsamples * self->info.bpf;
dst = (gint16 *) data;
for (i = 0; i < nsamples; i++) {
*dst = *samples;
samples += 2;
dst++;
}
hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
goto beach;
}
beach:
return length;
}
static guint
gst_wasapi_src_delay (GstAudioSrc * asrc)
{
/* FIXME: Implement */
return 0;
}
static void
gst_wasapi_src_reset (GstAudioSrc * asrc)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
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;
}
}
}
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);
if (G_UNLIKELY (hr != S_OK))
return 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;
}