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)
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;
"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");
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);
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);
+ /* FIXME: Implement */
+ return NULL;
+}
+
+static gboolean
+gst_wasapi_sink_open (GstAudioSink * asink)
+{
+ GstWasapiSink *self = GST_WASAPI_SINK (asink);
+ gboolean res = FALSE;
+ IAudioClient *client = NULL;
- if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
- *start = GST_BUFFER_TIMESTAMP (buffer);
+ 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;
+ }
- if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
- *end = *start + GST_BUFFER_DURATION (buffer);
- } else {
- *end = *start + self->buffer_time;
- }
+ self->client = client;
+ res = TRUE;
+
+beach:
+
+ return res;
+}
+
+static gboolean
+gst_wasapi_sink_close (GstAudioSink * asink)
+{
+ GstWasapiSink *self = GST_WASAPI_SINK (asink);
- *start += self->latency;
- *end += self->latency;
+ if (self->client != NULL) {
+ IUnknown_Release (self->client);
+ self->client = NULL;
}
+
+ return TRUE;
}
static gboolean
-gst_wasapi_sink_start (GstBaseSink * sink)
+gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
{
- GstWasapiSink *self = GST_WASAPI_SINK (sink);
+ GstWasapiSink *self = GST_WASAPI_SINK (asink);
gboolean res = FALSE;
- IAudioClient *client = NULL;
HRESULT hr;
+ REFERENCE_TIME latency_rt, def_period, min_period;
+ WAVEFORMATEXTENSIBLE format;
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))
+ 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_SetEventHandle (client, self->event_handle);
+ hr = IAudioClient_Initialize (self->client, AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ spec->buffer_time / 100, 0, (WAVEFORMATEX *) & format, NULL);
if (hr != S_OK) {
- GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
+ 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;
}
- hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
- (void **) &render_client);
+ 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::GetService "
- "(IID_IAudioRenderClient) failed");
+ GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
goto beach;
}
- hr = IAudioClient_Start (client);
+ 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->client = client;
self->render_client = render_client;
+ render_client = NULL;
res = TRUE;
beach:
- if (!res) {
- if (render_client != NULL)
- IUnknown_Release (render_client);
-
- if (client != NULL)
- IUnknown_Release (client);
- }
+ if (render_client != NULL)
+ IUnknown_Release (render_client);
return res;
}
static gboolean
-gst_wasapi_sink_stop (GstBaseSink * sink)
+gst_wasapi_sink_unprepare (GstAudioSink * asink)
{
- GstWasapiSink *self = GST_WASAPI_SINK (sink);
+ GstWasapiSink *self = GST_WASAPI_SINK (asink);
if (self->client != NULL) {
IAudioClient_Stop (self->client);
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);
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;
+ }
+ }
}
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);
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:
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)
projects / platform / upstream / gstreamer.git / commitdiff