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wasapi: Port wasapisink to GstAudioSink

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This commit is contained in:
Sebastian Dröge 2013-03-27 10:10:21 +01:00
parent 4b256e729e
commit 1445438a8b
7 changed files with 221 additions and 184 deletions
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3
sys/wasapi/Makefile.am
View file

@ -1,7 +1,6 @@
plugin_LTLIBRARIES = libgstwasapi.la
libgstwasapi_la_SOURCES = gstwasapi.c \
gstwasapisrc.c \
gstwasapisink.c \
gstwasapiutil.c
@ -12,7 +11,7 @@ libgstwasapi_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) \
libgstwasapi_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstwasapi_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstwasapisrc.h \
noinst_HEADERS = \
gstwasapisink.h \
gstwasapiutil.h

12
sys/wasapi/gstwasapi.c
View file

@ -21,21 +21,15 @@
# include <config.h>
#endif
#include "gstwasapisrc.h"
#include "gstwasapisink.h"
static gboolean
plugin_init (GstPlugin * plugin)
{
gboolean ret;
gst_element_register (plugin, "wasapisink", GST_RANK_NONE,
GST_TYPE_WASAPI_SINK);
ret = gst_element_register (plugin, "wasapisrc",
GST_RANK_NONE, GST_TYPE_WASAPI_SRC);
if (!ret)
return ret;
return gst_element_register (plugin, "wasapisink",
GST_RANK_NONE, GST_TYPE_WASAPI_SINK);
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,

266
sys/wasapi/gstwasapisink.c
View file

@ -45,19 +45,25 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S16LE, "
"layout = (string) interleaved, "
"rate = (int) 8000, " "channels = (int) 1"));
"rate = (int) 44100, " "channels = (int) 2"));
static void gst_wasapi_sink_dispose (GObject * object);
static void gst_wasapi_sink_finalize (GObject * object);
static void gst_wasapi_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end);
static gboolean gst_wasapi_sink_start (GstBaseSink * sink);
static gboolean gst_wasapi_sink_stop (GstBaseSink * sink);
static GstFlowReturn gst_wasapi_sink_render (GstBaseSink * sink,
GstBuffer * buffer);
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);
G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_BASE_SINK);
G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_SINK);
static void
gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
@ -65,6 +71,7 @@ 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;
@ -76,10 +83,15 @@ gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
"Stream audio to an audio capture device through WASAPI",
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
gstbasesink_class->get_times = gst_wasapi_sink_get_times;
gstbasesink_class->start = gst_wasapi_sink_start;
gstbasesink_class->stop = gst_wasapi_sink_stop;
gstbasesink_class->render = gst_wasapi_sink_render;
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");
@ -88,11 +100,6 @@ gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
static void
gst_wasapi_sink_init (GstWasapiSink * self)
{
self->rate = 8000;
self->buffer_time = 20 * GST_MSECOND;
self->period_time = 20 * GST_MSECOND;
self->latency = GST_CLOCK_TIME_NONE;
self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
CoInitialize (NULL);
@ -119,81 +126,125 @@ gst_wasapi_sink_finalize (GObject * object)
G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object);
}
static void
gst_wasapi_sink_get_times (GstBaseSink * sink,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end)
static GstCaps *
gst_wasapi_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
{
GstWasapiSink *self = GST_WASAPI_SINK (sink);
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
*start = GST_BUFFER_TIMESTAMP (buffer);
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
*end = *start + GST_BUFFER_DURATION (buffer);
} else {
*end = *start + self->buffer_time;
}
*start += self->latency;
*end += self->latency;
}
/* FIXME: Implement */
return NULL;
}
static gboolean
gst_wasapi_sink_start (GstBaseSink * sink)
gst_wasapi_sink_open (GstAudioSink * asink)
{
GstWasapiSink *self = GST_WASAPI_SINK (sink);
GstWasapiSink *self = GST_WASAPI_SINK (asink);
gboolean res = FALSE;
IAudioClient *client = NULL;
HRESULT hr;
IAudioRenderClient *render_client = NULL;
if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
FALSE, self->rate, self->buffer_time, self->period_time,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK, &client, &self->latency))
goto beach;
hr = IAudioClient_SetEventHandle (client, self->event_handle);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
goto beach;
}
hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
(void **) &render_client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService "
"(IID_IAudioRenderClient) failed");
goto beach;
}
hr = IAudioClient_Start (client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self), FALSE,
&client)) {
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("Failed to get default device"));
goto beach;
}
self->client = client;
self->render_client = render_client;
res = TRUE;
beach:
if (!res) {
if (render_client != NULL)
IUnknown_Release (render_client);
if (client != NULL)
IUnknown_Release (client);
}
return res;
}
static gboolean
gst_wasapi_sink_stop (GstBaseSink * sink)
gst_wasapi_sink_close (GstAudioSink * asink)
{
GstWasapiSink *self = GST_WASAPI_SINK (sink);
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, def_period, min_period;
WAVEFORMATEXTENSIBLE format;
IAudioRenderClient *render_client = NULL;
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_render_client (GST_ELEMENT (self), self->client,
&render_client)) {
goto beach;
}
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);
@ -204,36 +255,18 @@ gst_wasapi_sink_stop (GstBaseSink * sink)
self->render_client = NULL;
}
if (self->client != NULL) {
IUnknown_Release (self->client);
self->client = NULL;
}
return TRUE;
}
static GstFlowReturn
gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
static gint
gst_wasapi_sink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstWasapiSink *self = GST_WASAPI_SINK (sink);
GstFlowReturn ret = GST_FLOW_OK;
GstWasapiSink *self = GST_WASAPI_SINK (asink);
HRESULT hr;
GstMapInfo minfo;
const gint16 *src;
gint16 *dst = NULL;
guint nsamples;
guint i;
memset (&minfo, 0, sizeof (minfo));
if (!gst_buffer_map (buffer, &minfo, GST_MAP_READ)) {
GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
("Failed to map input buffer"));
ret = GST_FLOW_ERROR;
goto beach;
}
nsamples = minfo.size / sizeof (gint16);
nsamples = length / self->info.bpf;
WaitForSingleObject (self->event_handle, INFINITE);
@ -243,30 +276,51 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
("IAudioRenderClient::GetBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr)));
ret = GST_FLOW_ERROR;
length = 0;
goto beach;
}
src = (const gint16 *) minfo.data;
for (i = 0; i < nsamples; i++) {
dst[0] = *src;
dst[1] = *src;
src++;
dst += 2;
}
memcpy (dst, data, length);
hr = IAudioRenderClient_ReleaseBuffer (self->render_client, nsamples, 0);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioRenderClient::ReleaseBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
ret = GST_FLOW_ERROR;
length = 0;
goto beach;
}
beach:
if (minfo.data)
gst_buffer_unmap (buffer, &minfo);
return ret;
return length;
}
static guint
gst_wasapi_sink_delay (GstAudioSink * asink)
{
/* FIXME: Implement */
return 0;
}
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;
}
}
}

9
sys/wasapi/gstwasapisink.h
View file

@ -40,12 +40,9 @@ typedef struct _GstWasapiSinkClass GstWasapiSinkClass;
struct _GstWasapiSink
{
GstBaseSink base_sink;
GstAudioSink parent;
guint rate;
GstClockTime buffer_time;
GstClockTime period_time;
GstClockTime latency;
GstAudioInfo info;
IAudioClient * client;
IAudioRenderClient * render_client;
@ -54,7 +51,7 @@ struct _GstWasapiSink
struct _GstWasapiSinkClass
{
GstBaseSinkClass parent_class;
GstAudioSinkClass parent_class;
};
GType gst_wasapi_sink_get_type (void);

5
sys/wasapi/gstwasapisrc.c
View file

@ -180,7 +180,8 @@ gst_wasapi_src_start (GstBaseSrc * src)
&self->latency))
goto beach;
hr = IAudioClient_GetService (client, &IID_IAudioClock, (void**) &client_clock);
hr = IAudioClient_GetService (client, &IID_IAudioClock,
(void **) &client_clock);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
"failed");
@ -194,7 +195,7 @@ gst_wasapi_src_start (GstBaseSrc * src)
}
hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
(void**) &capture_client);
(void **) &capture_client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService "
"(IID_IAudioCaptureClient) failed");

95
sys/wasapi/gstwasapiutil.c
View file

@ -137,19 +137,13 @@ gst_wasapi_util_hresult_to_string (HRESULT hr)
gboolean
gst_wasapi_util_get_default_device_client (GstElement * element,
gboolean capture,
guint rate,
GstClockTime buffer_time,
GstClockTime period_time,
DWORD flags, IAudioClient ** ret_client, GstClockTime * ret_latency)
gboolean capture, IAudioClient ** ret_client)
{
gboolean res = FALSE;
HRESULT hr;
IMMDeviceEnumerator *enumerator = NULL;
IMMDevice *device = NULL;
IAudioClient *client = NULL;
REFERENCE_TIME latency_rt, def_period, min_period;
WAVEFORMATEXTENSIBLE format;
hr = CoCreateInstance (&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
&IID_IMMDeviceEnumerator, (void **) &enumerator);
@ -173,55 +167,9 @@ gst_wasapi_util_get_default_device_client (GstElement * element,
goto beach;
}
hr = IAudioClient_GetDevicePeriod (client, &def_period, &min_period);
if (hr != S_OK) {
GST_ERROR_OBJECT (element, "IAudioClient::GetDevicePeriod () failed");
goto beach;
}
ZeroMemory (&format, sizeof (format));
format.Format.cbSize = sizeof (format) - sizeof (format.Format);
format.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
format.Format.nChannels = 2;
format.Format.nSamplesPerSec = rate;
format.Format.wBitsPerSample = 16;
format.Format.nBlockAlign = format.Format.nChannels
* (format.Format.wBitsPerSample / 8);
format.Format.nAvgBytesPerSec = format.Format.nSamplesPerSec
* format.Format.nBlockAlign;
format.Samples.wValidBitsPerSample = 16;
format.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
format.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
hr = IAudioClient_Initialize (client, AUDCLNT_SHAREMODE_EXCLUSIVE, /* or AUDCLNT_SHAREMODE_SHARED */
flags, buffer_time / 100, /* buffer duration in 100s of ns */
period_time / 100, /* periodicity in 100s of ns */
(WAVEFORMATEX *) & format, NULL);
if (hr != S_OK) {
GST_ELEMENT_ERROR (element, RESOURCE, OPEN_READ, (NULL),
("IAudioClient::Initialize () failed: %s",
gst_wasapi_util_hresult_to_string (hr)));
goto beach;
}
hr = IAudioClient_GetStreamLatency (client, &latency_rt);
if (hr != S_OK) {
GST_ERROR_OBJECT (element, "IAudioClient::GetStreamLatency () failed");
goto beach;
}
GST_INFO_OBJECT (element, "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);
IUnknown_AddRef (client);
*ret_client = client;
*ret_latency = latency_rt * 100;
res = TRUE;
beach:
@ -237,6 +185,47 @@ beach:
return res;
}
gboolean
gst_wasapi_util_get_render_client (GstElement * element, IAudioClient * client,
IAudioRenderClient ** ret_render_client)
{
gboolean res = FALSE;
HRESULT hr;
IAudioRenderClient *render_client = NULL;
hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
(void **) &render_client);
if (hr != S_OK) {
GST_ERROR_OBJECT (element, "IAudioClient::GetService "
"(IID_IAudioRenderClient) failed");
goto beach;
}
*ret_render_client = render_client;
beach:
return res;
}
void
gst_wasapi_util_audio_info_to_waveformatex (GstAudioInfo * info,
WAVEFORMATEXTENSIBLE * format)
{
memset (format, 0, sizeof (*format));
format->Format.cbSize = sizeof (*format) - sizeof (format->Format);
format->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
format->Format.nChannels = info->channels;
format->Format.nSamplesPerSec = info->rate;
format->Format.wBitsPerSample = (info->bpf * 8) / format->Format.nChannels;
format->Format.nBlockAlign = info->bpf;
format->Format.nAvgBytesPerSec =
format->Format.nSamplesPerSec * format->Format.nBlockAlign;
format->Samples.wValidBitsPerSample = info->finfo->depth;
/* FIXME: Implement something here */
format->dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
format->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
}
#if 0
static WAVEFORMATEXTENSIBLE *
gst_wasapi_src_probe_device_format (GstWasapiSrc * self, IMMDevice * device)

15
sys/wasapi/gstwasapiutil.h
View file

@ -31,12 +31,15 @@ gst_wasapi_util_hresult_to_string (HRESULT hr);
gboolean
gst_wasapi_util_get_default_device_client (GstElement * element,
gboolean capture,
guint rate,
GstClockTime buffer_time,
GstClockTime period_time,
DWORD flags,
IAudioClient ** ret_client,
GstClockTime * ret_latency);
IAudioClient ** ret_client);
gboolean gst_wasapi_util_get_render_client (GstElement * element,
IAudioClient *client,
IAudioRenderClient ** ret_render_client);
void
gst_wasapi_util_audio_info_to_waveformatex (GstAudioInfo *info,
WAVEFORMATEXTENSIBLE *format);
#endif /* __GST_WASAPI_UTIL_H__ */