#endif
#include "gstwasapisrc.h"
-#include <gst/audio/gstaudioclock.h>
GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
#define GST_CAT_DEFAULT gst_wasapi_src_debug
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S16LE, "
"layout = (string) interleaved, "
- "rate = (int) 8000, " "channels = (int) 1"));
+ "rate = (int) 44100, " "channels = (int) 1"));
static void gst_wasapi_src_dispose (GObject * object);
static void gst_wasapi_src_finalize (GObject * object);
-static GstClock *gst_wasapi_src_provide_clock (GstElement * element);
+static GstCaps * gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
-static gboolean gst_wasapi_src_start (GstBaseSrc * src);
-static gboolean gst_wasapi_src_stop (GstBaseSrc * src);
-static gboolean gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query);
-
-static GstFlowReturn gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf);
+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_PUSH_SRC);
+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);
- GstPushSrcClass *gstpushsrc_class = GST_PUSH_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;
- gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
-
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
"Stream audio from an audio capture device through WASAPI",
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
- gstbasesrc_class->start = gst_wasapi_src_start;
- gstbasesrc_class->stop = gst_wasapi_src_stop;
- gstbasesrc_class->query = gst_wasapi_src_query;
+ gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_src_get_caps);
- gstpushsrc_class->create = gst_wasapi_src_create;
+ 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)
{
- GstBaseSrc *basesrc = GST_BASE_SRC (self);
-
- gst_base_src_set_format (basesrc, GST_FORMAT_TIME);
- gst_base_src_set_live (basesrc, TRUE);
+ /* override with a custom clock */
+ if (GST_AUDIO_BASE_SRC (self)->clock)
+ gst_object_unref (GST_AUDIO_BASE_SRC (self)->clock);
- self->rate = 8000;
- self->buffer_time = 20 * GST_MSECOND;
- self->period_time = 20 * GST_MSECOND;
- self->latency = GST_CLOCK_TIME_NONE;
- self->samples_per_buffer = self->rate / (GST_SECOND / self->period_time);
-
- self->start_time = GST_CLOCK_TIME_NONE;
- self->next_time = GST_CLOCK_TIME_NONE;
-
- self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
+ 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);
}
{
GstWasapiSrc *self = GST_WASAPI_SRC (object);
- if (self->clock != NULL) {
- gst_object_unref (self->clock);
- self->clock = NULL;
+ if (self->event_handle != NULL) {
+ CloseHandle (self->event_handle);
+ self->event_handle = NULL;
}
G_OBJECT_CLASS (gst_wasapi_src_parent_class)->dispose (object);
G_OBJECT_CLASS (gst_wasapi_src_parent_class)->finalize (object);
}
-static GstClock *
-gst_wasapi_src_provide_clock (GstElement * element)
+static GstCaps *
+gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
{
- GstWasapiSrc *self = GST_WASAPI_SRC (element);
- GstClock *clock;
+ /* TODO: Implement */
+ return NULL;
+}
+
+static gboolean
+gst_wasapi_src_open (GstAudioSrc * asrc)
+{
+ GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
+ gboolean res = FALSE;
+ IAudioClient * client = NULL;
- GST_OBJECT_LOCK (self);
+ 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;
+ }
- if (self->client_clock == NULL)
- goto wrong_state;
+ self->client = client;
+ res = TRUE;
- clock = GST_CLOCK (gst_object_ref (self->clock));
+beach:
- GST_OBJECT_UNLOCK (self);
- return clock;
+ return res;
+}
- /* ERRORS */
-wrong_state:
- {
- GST_OBJECT_UNLOCK (self);
- GST_DEBUG_OBJECT (self, "IAudioClock not acquired");
- return NULL;
+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_start (GstBaseSrc * src)
+gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
- GstWasapiSrc *self = GST_WASAPI_SRC (src);
+ GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
gboolean res = FALSE;
- IAudioClient *client = NULL;
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;
- if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
- TRUE, self->rate, self->buffer_time, self->period_time, 0, &client,
- &self->latency))
+ 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_GetService (client, &IID_IAudioClock,
- (void **) &client_clock);
+ hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
if (hr != S_OK) {
- GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
- "failed");
+ GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
+ goto beach;
+ }
+
+ if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client,
+ &client_clock)) {
goto beach;
}
goto beach;
}
- hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
- (void **) &capture_client);
- if (hr != S_OK) {
- GST_ERROR_OBJECT (self, "IAudioClient::GetService "
- "(IID_IAudioCaptureClient) failed");
+ if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client,
+ &capture_client)) {
goto beach;
}
- hr = IAudioClient_Start (client);
+ hr = IAudioClient_Start (self->client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
goto beach;
}
- self->client = client;
self->client_clock = client_clock;
self->client_clock_freq = client_clock_freq;
self->capture_client = capture_client;
if (client_clock != NULL)
IUnknown_Release (client_clock);
-
- if (client != NULL)
- IUnknown_Release (client);
}
return res;
}
static gboolean
-gst_wasapi_src_stop (GstBaseSrc * src)
+gst_wasapi_src_unprepare (GstAudioSrc * asrc)
{
- GstWasapiSrc *self = GST_WASAPI_SRC (src);
+ GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
if (self->client != NULL) {
IAudioClient_Stop (self->client);
self->client_clock = NULL;
}
- if (self->client != NULL) {
- IUnknown_Release (self->client);
- self->client = NULL;
- }
-
return TRUE;
}
-static gboolean
-gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
-{
- GstWasapiSrc *self = GST_WASAPI_SRC (src);
- gboolean ret = FALSE;
-
- GST_DEBUG_OBJECT (self, "query for %s",
- gst_query_type_get_name (GST_QUERY_TYPE (query)));
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_LATENCY:{
- GstClockTime min_latency, max_latency;
-
- min_latency = self->latency + self->period_time;
- max_latency = min_latency;
-
- GST_DEBUG_OBJECT (self, "reporting latency of min %" GST_TIME_FORMAT
- " max %" GST_TIME_FORMAT,
- GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
-
- gst_query_set_latency (query, TRUE, min_latency, max_latency);
- ret = TRUE;
- break;
- }
-
- default:
- ret =
- GST_BASE_SRC_CLASS (gst_wasapi_src_parent_class)->query (src, query);
- break;
- }
-
- return ret;
-}
-
-static GstFlowReturn
-gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
+static guint
+gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length,
+ GstClockTime * timestamp)
{
- GstWasapiSrc *self = GST_WASAPI_SRC (src);
- GstFlowReturn ret = GST_FLOW_OK;
- GstClock *clock;
- GstClockTime timestamp, duration = self->period_time;
+ GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
HRESULT hr;
gint16 *samples = NULL;
- guint32 nsamples_read = 0, nsamples;
+ guint32 nsamples = 0, length_samples;
DWORD flags = 0;
guint64 devpos;
guint i;
- GstMapInfo minfo;
gint16 *dst;
- GST_OBJECT_LOCK (self);
- clock = GST_ELEMENT_CLOCK (self);
- if (clock != NULL)
- gst_object_ref (clock);
- GST_OBJECT_UNLOCK (self);
-
- if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
- GstClockID id;
-
- id = gst_clock_new_single_shot_id (clock, self->next_time);
- gst_clock_id_wait (id, NULL);
- gst_clock_id_unref (id);
- }
+ WaitForSingleObject (self->event_handle, INFINITE);
do {
hr = IAudioCaptureClient_GetBuffer (self->capture_client,
- (BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
+ (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));
- ret = GST_FLOW_ERROR;
+ length = 0;
goto beach;
}
devpos, (guint) flags);
}
- /* FIXME: Why do we get 1024 sometimes and not a multiple of
- * samples_per_buffer? Shouldn't WASAPI provide a DISCONT
- * flag if we read too slow?
- */
- nsamples = nsamples_read;
- g_assert (nsamples >= self->samples_per_buffer);
- if (nsamples > self->samples_per_buffer) {
- GST_WARNING_OBJECT (self,
- "devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
- devpos, nsamples, self->samples_per_buffer);
-
- nsamples = self->samples_per_buffer;
- }
-
- if (clock == NULL || clock == self->clock) {
- timestamp =
- gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
- } else {
- GstClockTime base_time;
+ length_samples = length / self->info.bpf;
+ nsamples = MIN (length_samples, nsamples);
+ length = nsamples * self->info.bpf;
- timestamp = gst_clock_get_time (clock);
-
- base_time = GST_ELEMENT_CAST (self)->base_time;
- if (timestamp > base_time)
- timestamp -= base_time;
- else
- timestamp = 0;
-
- if (timestamp > duration)
- timestamp -= duration;
- else
- timestamp = 0;
- }
-
- *buf = gst_buffer_new_and_alloc (nsamples * sizeof (gint16));
-
- GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
- GST_BUFFER_TIMESTAMP (*buf) = timestamp;
- GST_BUFFER_DURATION (*buf) = duration;
-
- gst_buffer_map (*buf, &minfo, GST_MAP_WRITE);
- dst = (gint16 *) minfo.data;
+ dst = (gint16 *) data;
for (i = 0; i < nsamples; i++) {
*dst = *samples;
samples += 2;
dst++;
}
- gst_buffer_unmap (*buf, &minfo);
- hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
+ 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));
- ret = GST_FLOW_ERROR;
goto beach;
}
beach:
- if (clock != NULL)
- gst_object_unref (clock);
- return ret;
+ 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
projects / platform / upstream / gstreamer.git / commitdiff