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-rw-r--r--sys/wasapi2/gstwasapi2client.cpp1137
-rw-r--r--sys/wasapi2/gstwasapi2client.h44
-rw-r--r--sys/wasapi2/gstwasapi2device.c2
-rw-r--r--sys/wasapi2/gstwasapi2ringbuffer.cpp1015
-rw-r--r--sys/wasapi2/gstwasapi2ringbuffer.h55
-rw-r--r--sys/wasapi2/gstwasapi2sink.c391
-rw-r--r--sys/wasapi2/gstwasapi2sink.h2
-rw-r--r--sys/wasapi2/gstwasapi2src.c388
-rw-r--r--sys/wasapi2/gstwasapi2src.h2
-rw-r--r--sys/wasapi2/gstwasapi2util.c237
-rw-r--r--sys/wasapi2/gstwasapi2util.h18
-rw-r--r--sys/wasapi2/meson.build4
-rw-r--r--sys/wasapi2/plugin.c18
13 files changed, 1629 insertions, 1684 deletions
diff --git a/sys/wasapi2/gstwasapi2client.cpp b/sys/wasapi2/gstwasapi2client.cpp
index b7555e4d5..c27d6f8ff 100644
--- a/sys/wasapi2/gstwasapi2client.cpp
+++ b/sys/wasapi2/gstwasapi2client.cpp
@@ -41,51 +41,6 @@
#include <locale>
#include <codecvt>
-/* Desktop only defines */
-#ifndef KSAUDIO_SPEAKER_MONO
-#define KSAUDIO_SPEAKER_MONO (SPEAKER_FRONT_CENTER)
-#endif
-#ifndef KSAUDIO_SPEAKER_1POINT1
-#define KSAUDIO_SPEAKER_1POINT1 (SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY)
-#endif
-#ifndef KSAUDIO_SPEAKER_STEREO
-#define KSAUDIO_SPEAKER_STEREO (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT)
-#endif
-#ifndef KSAUDIO_SPEAKER_2POINT1
-#define KSAUDIO_SPEAKER_2POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY)
-#endif
-#ifndef KSAUDIO_SPEAKER_3POINT0
-#define KSAUDIO_SPEAKER_3POINT0 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER)
-#endif
-#ifndef KSAUDIO_SPEAKER_3POINT1
-#define KSAUDIO_SPEAKER_3POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
- SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY)
-#endif
-#ifndef KSAUDIO_SPEAKER_QUAD
-#define KSAUDIO_SPEAKER_QUAD (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
- SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
-#endif
-#define KSAUDIO_SPEAKER_SURROUND (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
- SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER)
-#ifndef KSAUDIO_SPEAKER_5POINT0
-#define KSAUDIO_SPEAKER_5POINT0 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
- SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
-#endif
-#define KSAUDIO_SPEAKER_5POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
- SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
- SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
-#ifndef KSAUDIO_SPEAKER_7POINT0
-#define KSAUDIO_SPEAKER_7POINT0 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
- SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
- SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
-#endif
-#ifndef KSAUDIO_SPEAKER_7POINT1
-#define KSAUDIO_SPEAKER_7POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
- SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
- SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
- SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER)
-#endif
-
/* *INDENT-OFF* */
using namespace ABI::Windows::ApplicationModel::Core;
using namespace ABI::Windows::Foundation;
@@ -107,8 +62,7 @@ G_END_DECLS
static void
gst_wasapi2_client_on_device_activated (GstWasapi2Client * client,
- IAudioClient3 * audio_client);
-
+ IAudioClient * audio_client);
/* *INDENT-OFF* */
class GstWasapiDeviceActivator
@@ -150,7 +104,7 @@ public:
STDMETHOD(ActivateCompleted)
(IActivateAudioInterfaceAsyncOperation *async_op)
{
- ComPtr<IAudioClient3> audio_client;
+ ComPtr<IAudioClient> audio_client;
HRESULT hr = S_OK;
HRESULT hr_async_op = S_OK;
ComPtr<IUnknown> audio_interface;
@@ -255,6 +209,7 @@ private:
ComPtr<ICoreDispatcher> dispatcher_;
};
/* *INDENT-ON* */
+
typedef enum
{
GST_WASAPI2_CLIENT_ACTIVATE_FAILED = -1,
@@ -270,47 +225,27 @@ enum
PROP_DEVICE_NAME,
PROP_DEVICE_INDEX,
PROP_DEVICE_CLASS,
- PROP_LOW_LATENCY,
PROP_DISPATCHER,
};
#define DEFAULT_DEVICE_INDEX -1
#define DEFAULT_DEVICE_CLASS GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE
-#define DEFAULT_LOW_LATENCY FALSE
struct _GstWasapi2Client
{
GstObject parent;
GstWasapi2ClientDeviceClass device_class;
- gboolean low_latency;
gchar *device_id;
gchar *device_name;
gint device_index;
gpointer dispatcher;
- IAudioClient3 *audio_client;
- IAudioCaptureClient *audio_capture_client;
- IAudioRenderClient *audio_render_client;
- ISimpleAudioVolume *audio_volume;
+ IAudioClient *audio_client;
GstWasapiDeviceActivator *activator;
- WAVEFORMATEX *mix_format;
GstCaps *supported_caps;
- HANDLE event_handle;
- HANDLE cancellable;
- gboolean opened;
- gboolean running;
-
- guint32 device_period;
- guint32 buffer_frame_count;
-
- GstAudioChannelPosition *positions;
-
- /* Used for capture mode */
- GstAdapter *adapter;
-
GThread *thread;
GMutex lock;
GCond cond;
@@ -330,7 +265,7 @@ gst_wasapi2_client_device_class_get_type (void)
static const GEnumValue types[] = {
{GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE, "Capture", "capture"},
{GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER, "Render", "render"},
- {0, NULL, NULL}
+ {0, nullptr, nullptr}
};
if (g_once_init_enter (&class_type)) {
@@ -342,7 +277,6 @@ gst_wasapi2_client_device_class_get_type (void)
}
static void gst_wasapi2_client_constructed (GObject * object);
-static void gst_wasapi2_client_dispose (GObject * object);
static void gst_wasapi2_client_finalize (GObject * object);
static void gst_wasapi2_client_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
@@ -365,17 +299,16 @@ gst_wasapi2_client_class_init (GstWasapi2ClientClass * klass)
G_PARAM_STATIC_STRINGS);
gobject_class->constructed = gst_wasapi2_client_constructed;
- gobject_class->dispose = gst_wasapi2_client_dispose;
gobject_class->finalize = gst_wasapi2_client_finalize;
gobject_class->get_property = gst_wasapi2_client_get_property;
gobject_class->set_property = gst_wasapi2_client_set_property;
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
- "WASAPI playback device as a GUID string", NULL, param_flags));
+ "WASAPI playback device as a GUID string", nullptr, param_flags));
g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device Name",
- "The human-readable device name", NULL, param_flags));
+ "The human-readable device name", nullptr, param_flags));
g_object_class_install_property (gobject_class, PROP_DEVICE_INDEX,
g_param_spec_int ("device-index", "Device Index",
"The zero-based device index", -1, G_MAXINT, DEFAULT_DEVICE_INDEX,
@@ -384,10 +317,6 @@ gst_wasapi2_client_class_init (GstWasapi2ClientClass * klass)
g_param_spec_enum ("device-class", "Device Class",
"Device class", GST_TYPE_WASAPI2_CLIENT_DEVICE_CLASS,
DEFAULT_DEVICE_CLASS, param_flags));
- g_object_class_install_property (gobject_class, PROP_LOW_LATENCY,
- g_param_spec_boolean ("low-latency", "Low latency",
- "Optimize all settings for lowest latency. Always safe to enable.",
- DEFAULT_LOW_LATENCY, param_flags));
g_object_class_install_property (gobject_class, PROP_DISPATCHER,
g_param_spec_pointer ("dispatcher", "Dispatcher",
"ICoreDispatcher COM object to use", param_flags));
@@ -398,11 +327,6 @@ gst_wasapi2_client_init (GstWasapi2Client * self)
{
self->device_index = DEFAULT_DEVICE_INDEX;
self->device_class = DEFAULT_DEVICE_CLASS;
- self->low_latency = DEFAULT_LOW_LATENCY;
-
- self->adapter = gst_adapter_new ();
- self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
- self->cancellable = CreateEvent (NULL, TRUE, FALSE, NULL);
g_mutex_init (&self->lock);
g_cond_init (&self->cond);
@@ -438,44 +362,26 @@ gst_wasapi2_client_constructed (GObject * object)
}
static void
-gst_wasapi2_client_dispose (GObject * object)
+gst_wasapi2_client_finalize (GObject * object)
{
GstWasapi2Client *self = GST_WASAPI2_CLIENT (object);
- GST_DEBUG_OBJECT (self, "dispose");
-
- gst_clear_caps (&self->supported_caps);
-
if (self->loop) {
g_main_loop_quit (self->loop);
g_thread_join (self->thread);
g_main_context_unref (self->context);
g_main_loop_unref (self->loop);
- self->thread = NULL;
- self->context = NULL;
- self->loop = NULL;
+ self->thread = nullptr;
+ self->context = nullptr;
+ self->loop = nullptr;
}
- g_clear_object (&self->adapter);
-
- G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-static void
-gst_wasapi2_client_finalize (GObject * object)
-{
- GstWasapi2Client *self = GST_WASAPI2_CLIENT (object);
+ gst_clear_caps (&self->supported_caps);
g_free (self->device_id);
g_free (self->device_name);
- g_free (self->positions);
-
- CoTaskMemFree (self->mix_format);
- CloseHandle (self->event_handle);
- CloseHandle (self->cancellable);
-
g_mutex_clear (&self->lock);
g_cond_clear (&self->cond);
@@ -504,9 +410,6 @@ gst_wasapi2_client_get_property (GObject * object, guint prop_id,
case PROP_DEVICE_CLASS:
g_value_set_enum (value, self->device_class);
break;
- case PROP_LOW_LATENCY:
- g_value_set_boolean (value, self->low_latency);
- break;
case PROP_DISPATCHER:
g_value_set_pointer (value, self->dispatcher);
break;
@@ -538,9 +441,6 @@ gst_wasapi2_client_set_property (GObject * object, guint prop_id,
self->device_class =
(GstWasapi2ClientDeviceClass) g_value_get_enum (value);
break;
- case PROP_LOW_LATENCY:
- self->low_latency = g_value_get_boolean (value);
- break;
case PROP_DISPATCHER:
self->dispatcher = g_value_get_pointer (value);
break;
@@ -564,7 +464,7 @@ gst_wasapi2_client_main_loop_running_cb (GstWasapi2Client * self)
static void
gst_wasapi2_client_on_device_activated (GstWasapi2Client * self,
- IAudioClient3 * audio_client)
+ IAudioClient * audio_client)
{
GST_INFO_OBJECT (self, "Device activated");
@@ -912,7 +812,7 @@ run_loop:
source = g_idle_source_new ();
g_source_set_callback (source,
- (GSourceFunc) gst_wasapi2_client_main_loop_running_cb, self, NULL);
+ (GSourceFunc) gst_wasapi2_client_main_loop_running_cb, self, nullptr);
g_source_attach (source, self->context);
g_source_unref (source);
@@ -922,32 +822,7 @@ run_loop:
g_main_context_pop_thread_default (self->context);
- gst_wasapi2_client_stop (self);
-
- if (self->audio_volume) {
- /* this mute state seems to be global setting for this device
- * Explicitly disable mute for later use of this audio device
- * by other application. Otherwise users would blame GStreamer
- * if we close audio device with muted state */
- self->audio_volume->SetMute (FALSE, nullptr);
- self->audio_volume->Release ();
- self->audio_volume = NULL;
- }
-
- if (self->audio_render_client) {
- self->audio_render_client->Release ();
- self->audio_render_client = NULL;
- }
-
- if (self->audio_capture_client) {
- self->audio_capture_client->Release ();
- self->audio_capture_client = NULL;
- }
-
- if (self->audio_client) {
- self->audio_client->Release ();
- self->audio_client = NULL;
- }
+ GST_WASAPI2_CLEAR_COM (self->audio_client);
/* Reset explicitly to ensure that it happens before
* RoInitializeWrapper dtor is called */
@@ -955,986 +830,48 @@ run_loop:
GST_DEBUG_OBJECT (self, "Exit thread function");
- return NULL;
-}
-
-static const gchar *
-gst_waveformatex_to_audio_format (WAVEFORMATEXTENSIBLE * format)
-{
- const gchar *fmt_str = NULL;
- GstAudioFormat fmt = GST_AUDIO_FORMAT_UNKNOWN;
-
- if (format->Format.wFormatTag == WAVE_FORMAT_PCM) {
- fmt = gst_audio_format_build_integer (TRUE, G_LITTLE_ENDIAN,
- format->Format.wBitsPerSample, format->Format.wBitsPerSample);
- } else if (format->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT) {
- if (format->Format.wBitsPerSample == 32)
- fmt = GST_AUDIO_FORMAT_F32LE;
- else if (format->Format.wBitsPerSample == 64)
- fmt = GST_AUDIO_FORMAT_F64LE;
- } else if (format->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
- if (IsEqualGUID (format->SubFormat, KSDATAFORMAT_SUBTYPE_PCM)) {
- fmt = gst_audio_format_build_integer (TRUE, G_LITTLE_ENDIAN,
- format->Format.wBitsPerSample, format->Samples.wValidBitsPerSample);
- } else if (IsEqualGUID (format->SubFormat, KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) {
- if (format->Format.wBitsPerSample == 32
- && format->Samples.wValidBitsPerSample == 32)
- fmt = GST_AUDIO_FORMAT_F32LE;
- else if (format->Format.wBitsPerSample == 64 &&
- format->Samples.wValidBitsPerSample == 64)
- fmt = GST_AUDIO_FORMAT_F64LE;
- }
- }
-
- if (fmt != GST_AUDIO_FORMAT_UNKNOWN)
- fmt_str = gst_audio_format_to_string (fmt);
-
- return fmt_str;
-}
-
-static void
-gst_wasapi_util_channel_position_all_none (guint channels,
- GstAudioChannelPosition * position)
-{
- int ii;
- for (ii = 0; ii < channels; ii++)
- position[ii] = GST_AUDIO_CHANNEL_POSITION_NONE;
-}
-
-static struct
-{
- guint64 wasapi_pos;
- GstAudioChannelPosition gst_pos;
-} wasapi_to_gst_pos[] = {
- {SPEAKER_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT},
- {SPEAKER_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
- {SPEAKER_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER},
- {SPEAKER_LOW_FREQUENCY, GST_AUDIO_CHANNEL_POSITION_LFE1},
- {SPEAKER_BACK_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT},
- {SPEAKER_BACK_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
- {SPEAKER_FRONT_LEFT_OF_CENTER,
- GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER},
- {SPEAKER_FRONT_RIGHT_OF_CENTER,
- GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
- {SPEAKER_BACK_CENTER, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
- /* Enum values diverge from this point onwards */
- {SPEAKER_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT},
- {SPEAKER_SIDE_RIGHT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT},
- {SPEAKER_TOP_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER},
- {SPEAKER_TOP_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT},
- {SPEAKER_TOP_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER},
- {SPEAKER_TOP_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT},
- {SPEAKER_TOP_BACK_LEFT, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT},
- {SPEAKER_TOP_BACK_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER},
- {SPEAKER_TOP_BACK_RIGHT, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
-};
-
-static DWORD default_ch_masks[] = {
- 0,
- KSAUDIO_SPEAKER_MONO,
- /* 2ch */
- KSAUDIO_SPEAKER_STEREO,
- /* 2.1ch */
- /* KSAUDIO_SPEAKER_3POINT0 ? */
- KSAUDIO_SPEAKER_2POINT1,
- /* 4ch */
- /* KSAUDIO_SPEAKER_3POINT1 or KSAUDIO_SPEAKER_SURROUND ? */
- KSAUDIO_SPEAKER_QUAD,
- /* 5ch */
- KSAUDIO_SPEAKER_5POINT0,
- /* 5.1ch */
- KSAUDIO_SPEAKER_5POINT1,
- /* 7ch */
- KSAUDIO_SPEAKER_7POINT0,
- /* 7.1ch */
- KSAUDIO_SPEAKER_7POINT1,
-};
-
-/* Parse WAVEFORMATEX to get the gstreamer channel mask, and the wasapi channel
- * positions so GstAudioRingbuffer can reorder the audio data to match the
- * gstreamer channel order. */
-static guint64
-gst_wasapi2_util_waveformatex_to_channel_mask (WORD nChannels,
- DWORD dwChannelMask, GstAudioChannelPosition ** out_position)
-{
- int ii, ch;
- guint64 mask = 0;
- GstAudioChannelPosition *pos = NULL;
-
- pos = g_new (GstAudioChannelPosition, nChannels);
- gst_wasapi_util_channel_position_all_none (nChannels, pos);
-
- /* Too many channels, have to assume that they are all non-positional */
- if (nChannels > G_N_ELEMENTS (wasapi_to_gst_pos)) {
- GST_INFO ("Got too many (%i) channels, assuming non-positional", nChannels);
- goto out;
- }
-
- /* Too many bits in the channel mask, and the bits don't match nChannels */
- if (dwChannelMask >> (G_N_ELEMENTS (wasapi_to_gst_pos) + 1) != 0) {
- GST_WARNING ("Too many bits in channel mask (%lu), assuming "
- "non-positional", dwChannelMask);
- goto out;
- }
-
- /* Map WASAPI's channel mask to Gstreamer's channel mask and positions.
- * If the no. of bits in the mask > nChannels, we will ignore the extra. */
- for (ii = 0, ch = 0; ii < G_N_ELEMENTS (wasapi_to_gst_pos) && ch < nChannels;
- ii++) {
- if (!(dwChannelMask & wasapi_to_gst_pos[ii].wasapi_pos))
- /* no match, try next */
- continue;
- mask |= G_GUINT64_CONSTANT (1) << wasapi_to_gst_pos[ii].gst_pos;
- pos[ch++] = wasapi_to_gst_pos[ii].gst_pos;
- }
-
- /* XXX: Warn if some channel masks couldn't be mapped? */
-
- GST_DEBUG ("Converted WASAPI mask 0x%" G_GINT64_MODIFIER "x -> 0x%"
- G_GINT64_MODIFIER "x", (guint64) dwChannelMask, (guint64) mask);
-
-out:
- if (out_position)
- *out_position = pos;
- return mask;
-}
-
-static gboolean
-gst_wasapi2_util_parse_waveformatex (WAVEFORMATEXTENSIBLE * format,
- GstCaps * template_caps, GstCaps ** out_caps,
- GstAudioChannelPosition ** out_positions)
-{
- const gchar *afmt;
- guint64 channel_mask = 0;
- DWORD dwChannelMask = 0;
- WORD nChannels;
-
- *out_caps = NULL;
-
- /* TODO: handle SPDIF and other encoded formats */
-
- /* 1 or 2 channels <= 16 bits sample size OR
- * 1 or 2 channels > 16 bits sample size or >2 channels */
- if (format->Format.wFormatTag != WAVE_FORMAT_PCM &&
- format->Format.wFormatTag != WAVE_FORMAT_IEEE_FLOAT &&
- format->Format.wFormatTag != WAVE_FORMAT_EXTENSIBLE)
- /* Unhandled format tag */
- return FALSE;
-
- /* WASAPI can only tell us one canonical mix format that it will accept. The
- * alternative is calling IsFormatSupported on all combinations of formats.
- * Instead, it's simpler and faster to require conversion inside gstreamer */
- afmt = gst_waveformatex_to_audio_format (format);
- if (afmt == NULL)
- return FALSE;
-
- *out_caps = gst_caps_copy (template_caps);
-
- nChannels = format->Format.nChannels;
- if (format->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
- dwChannelMask = format->dwChannelMask;
- }
-
- if (nChannels > 2 && !dwChannelMask) {
- GST_WARNING ("Unknown channel mask value for %d channel stream",
- format->Format.nChannels);
- if (nChannels >= G_N_ELEMENTS (default_ch_masks)) {
- GST_ERROR ("To may channels %d", nChannels);
- return FALSE;
- }
-
- dwChannelMask = default_ch_masks[nChannels];
- }
-
- channel_mask =
- gst_wasapi2_util_waveformatex_to_channel_mask (nChannels,
- dwChannelMask, out_positions);
-
- gst_caps_set_simple (*out_caps,
- "format", G_TYPE_STRING, afmt,
- "channels", G_TYPE_INT, format->Format.nChannels,
- "rate", G_TYPE_INT, format->Format.nSamplesPerSec, NULL);
-
- if (channel_mask) {
- gst_caps_set_simple (*out_caps,
- "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
- }
-
- return TRUE;
+ return nullptr;
}
GstCaps *
gst_wasapi2_client_get_caps (GstWasapi2Client * client)
{
- WAVEFORMATEX *format = NULL;
+ WAVEFORMATEX *mix_format = nullptr;
static GstStaticCaps static_caps = GST_STATIC_CAPS (GST_WASAPI2_STATIC_CAPS);
GstCaps *scaps;
HRESULT hr;
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), NULL);
+ g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), nullptr);
if (client->supported_caps)
return gst_caps_ref (client->supported_caps);
if (!client->audio_client) {
GST_WARNING_OBJECT (client, "IAudioClient3 wasn't configured");
- return NULL;
+ return nullptr;
}
- CoTaskMemFree (client->mix_format);
- client->mix_format = nullptr;
-
- g_clear_pointer (&client->positions, g_free);
-
- hr = client->audio_client->GetMixFormat (&format);
- if (!gst_wasapi2_result (hr))
- return NULL;
+ hr = client->audio_client->GetMixFormat (&mix_format);
+ if (!gst_wasapi2_result (hr)) {
+ GST_WARNING_OBJECT (client, "Failed to get mix format");
+ return nullptr;
+ }
scaps = gst_static_caps_get (&static_caps);
- gst_wasapi2_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
- scaps, &client->supported_caps, &client->positions);
+ gst_wasapi2_util_parse_waveformatex (mix_format,
+ scaps, &client->supported_caps, nullptr);
gst_caps_unref (scaps);
- client->mix_format = format;
+ CoTaskMemFree (mix_format);
if (!client->supported_caps) {
GST_ERROR_OBJECT (client, "No caps from subclass");
- return NULL;
+ return nullptr;
}
return gst_caps_ref (client->supported_caps);
}
-static HRESULT
-gst_wasapi2_client_initialize_audio_client3 (GstWasapi2Client * self)
-{
- HRESULT hr = S_OK;
- UINT32 default_period, fundamental_period, min_period, max_period;
- /* AUDCLNT_STREAMFLAGS_NOPERSIST is not allowed for
- * InitializeSharedAudioStream */
- DWORD stream_flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
- WAVEFORMATEX *format = NULL;
- UINT32 period;
- IAudioClient3 *audio_client = self->audio_client;
-
- hr = audio_client->GetSharedModeEnginePeriod (self->mix_format,
- &default_period, &fundamental_period, &min_period, &max_period);
- if (!gst_wasapi2_result (hr))
- goto done;
-
- GST_INFO_OBJECT (self, "Using IAudioClient3, default period %d frames, "
- "fundamental period %d frames, minimum period %d frames, maximum period "
- "%d frames", default_period, fundamental_period, min_period, max_period);
-
- hr = audio_client->InitializeSharedAudioStream (stream_flags, min_period,
- self->mix_format, nullptr);
-
- if (!gst_wasapi2_result (hr)) {
- GST_WARNING_OBJECT (self, "Failed to initialize IAudioClient3");
- goto done;
- }
-
- /* query period again to be ensured */
- hr = audio_client->GetCurrentSharedModeEnginePeriod (&format, &period);
- if (!gst_wasapi2_result (hr)) {
- GST_WARNING_OBJECT (self, "Failed to get current period");
- goto done;
- }
-
- self->device_period = period;
-
-done:
- CoTaskMemFree (format);
-
- return hr;
-}
-
-static void
-gst_wasapi2_util_get_best_buffer_sizes (GstAudioRingBufferSpec * spec,
- REFERENCE_TIME default_period, REFERENCE_TIME min_period,
- REFERENCE_TIME * ret_period, REFERENCE_TIME * ret_buffer_duration)
-{
- REFERENCE_TIME use_period, use_buffer;
-
- /* Shared mode always runs at the default period, so if we want a larger
- * period (for lower CPU usage), we do it as a multiple of that */
- use_period = default_period;
-
- /* Ensure that the period (latency_time) used is an integral multiple of
- * either the default period or the minimum period */
- use_period = use_period * MAX ((spec->latency_time * 10) / use_period, 1);
-
- /* Ask WASAPI to create a software ringbuffer of at least this size; it may
- * be larger so the actual buffer time may be different, which is why after
- * initialization we read the buffer duration actually in-use and set
- * segsize/segtotal from that. */
- use_buffer = spec->buffer_time * 10;
- /* Has to be at least twice the period */
- if (use_buffer < 2 * use_period)
- use_buffer = 2 * use_period;
-
- *ret_period = use_period;
- *ret_buffer_duration = use_buffer;
-}
-
-static HRESULT
-gst_wasapi2_client_initialize_audio_client (GstWasapi2Client * self,
- GstAudioRingBufferSpec * spec)
-{
- REFERENCE_TIME default_period, min_period;
- REFERENCE_TIME device_period, device_buffer_duration;
- guint rate;
- DWORD stream_flags =
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
- HRESULT hr;
- IAudioClient3 *audio_client = self->audio_client;
-
- hr = audio_client->GetDevicePeriod (&default_period, &min_period);
- if (!gst_wasapi2_result (hr)) {
- GST_WARNING_OBJECT (self, "Couldn't get device period info");
- return hr;
- }
-
- GST_INFO_OBJECT (self, "wasapi2 default period: %" G_GINT64_FORMAT
- ", min period: %" G_GINT64_FORMAT, default_period, min_period);
-
- rate = GST_AUDIO_INFO_RATE (&spec->info);
-
- if (self->low_latency) {
- device_period = default_period;
- /* this should be same as hnsPeriodicity
- * when AUDCLNT_STREAMFLAGS_EVENTCALLBACK is used
- * And in case of shared mode, hnsPeriodicity should be zero, so
- * this value should be zero as well */
- device_buffer_duration = 0;
- } else {
- /* Clamp values to integral multiples of an appropriate period */
- gst_wasapi2_util_get_best_buffer_sizes (spec,
- default_period, min_period, &device_period, &device_buffer_duration);
- }
-
- hr = audio_client->Initialize (AUDCLNT_SHAREMODE_SHARED, stream_flags,
- device_buffer_duration,
- /* This must always be 0 in shared mode */
- 0, self->mix_format, nullptr);
- if (!gst_wasapi2_result (hr)) {
- GST_WARNING_OBJECT (self, "Couldn't initialize audioclient");
- return hr;
- }
-
- /* device_period can be a non-power-of-10 value so round while converting */
- self->device_period =
- gst_util_uint64_scale_round (device_period, rate * 100, GST_SECOND);
-
- return S_OK;
-}
-
-HRESULT
-gst_wasapi2_client_open (GstWasapi2Client * client,
- GstAudioRingBufferSpec * spec, GstAudioRingBuffer * buf)
-{
- HRESULT hr = E_FAIL;
- REFERENCE_TIME latency_rt;
- guint bpf, rate;
- IAudioClient3 *audio_client;
- /* *INDENT-OFF* */
- ComPtr<ISimpleAudioVolume> audio_volume;
- /* *INDENT-ON* */
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
-
- /* FIXME: Once IAudioClient3 was initialized, we may need to re-open
- * IAudioClient3 in order to handle audio format change */
- if (client->opened) {
- GST_INFO_OBJECT (client, "IAudioClient3 object is initialized already");
- return S_OK;
- }
-
- audio_client = client->audio_client;
-
- if (!audio_client) {
- GST_ERROR_OBJECT (client, "IAudioClient3 object wasn't configured");
- return E_FAIL;
- }
-
- if (!client->mix_format) {
- GST_ERROR_OBJECT (client, "Unknown mix format");
- return E_FAIL;
- }
-
- /* Only use audioclient3 when low-latency is requested because otherwise
- * very slow machines and VMs with 1 CPU allocated will get glitches:
- * https://bugzilla.gnome.org/show_bug.cgi?id=794497 */
- if (client->low_latency)
- hr = gst_wasapi2_client_initialize_audio_client3 (client);
-
- /* Try again if IAudioClinet3 API is unavailable.
- * NOTE: IAudioClinet3:: methods might not be available for default device
- * NOTE: The default device is a special device which is needed for supporting
- * automatic stream routing
- * https://docs.microsoft.com/en-us/windows/win32/coreaudio/automatic-stream-routing
- */
- if (FAILED (hr))
- hr = gst_wasapi2_client_initialize_audio_client (client, spec);
-
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client, "Failed to initialize audioclient");
- return hr;
- }
-
- bpf = GST_AUDIO_INFO_BPF (&spec->info);
- rate = GST_AUDIO_INFO_RATE (&spec->info);
-
- /* Total size in frames of the allocated buffer that we will read from */
- hr = audio_client->GetBufferSize (&client->buffer_frame_count);
- if (!gst_wasapi2_result (hr))
- return hr;
-
- GST_INFO_OBJECT (client, "buffer size is %i frames, device period is %i "
- "frames, bpf is %i bytes, rate is %i Hz", client->buffer_frame_count,
- client->device_period, bpf, rate);
-
- /* Actual latency-time/buffer-time will be different now */
- spec->segsize = client->device_period * bpf;
-
- /* We need a minimum of 2 segments to ensure glitch-free playback */
- spec->segtotal = MAX (client->buffer_frame_count * bpf / spec->segsize, 2);
-
- GST_INFO_OBJECT (client, "segsize is %i, segtotal is %i", spec->segsize,
- spec->segtotal);
-
- /* Get WASAPI latency for logging */
- hr = audio_client->GetStreamLatency (&latency_rt);
- if (!gst_wasapi2_result (hr))
- return hr;
-
- GST_INFO_OBJECT (client, "wasapi2 stream latency: %" G_GINT64_FORMAT " (%"
- G_GINT64_FORMAT " ms)", latency_rt, latency_rt / 10000);
-
- /* Set the event handler which will trigger read/write */
- hr = audio_client->SetEventHandle (client->event_handle);
- if (!gst_wasapi2_result (hr))
- return hr;
-
- if (client->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER) {
- /* *INDENT-OFF* */
- ComPtr<IAudioRenderClient> render_client;
- /* *INDENT-ON* */
-
- hr = audio_client->GetService (IID_PPV_ARGS (&render_client));
- if (!gst_wasapi2_result (hr))
- return hr;
-
- client->audio_render_client = render_client.Detach ();
- } else {
- /* *INDENT-OFF* */
- ComPtr<IAudioCaptureClient> capture_client;
- /* *INDENT-ON* */
-
- hr = audio_client->GetService (IID_PPV_ARGS (&capture_client));
- if (!gst_wasapi2_result (hr))
- return hr;
-
- client->audio_capture_client = capture_client.Detach ();
- }
-
- hr = audio_client->GetService (IID_PPV_ARGS (&audio_volume));
- if (!gst_wasapi2_result (hr))
- return hr;
-
- client->audio_volume = audio_volume.Detach ();
- client->audio_volume->SetMute (FALSE, nullptr);
-
- gst_audio_ring_buffer_set_channel_positions (buf, client->positions);
-
- client->opened = TRUE;
-
- return S_OK;
-}
-
-/* Get the empty space in the buffer that we have to write to */
-static HRESULT
-gst_wasapi2_client_get_can_frames (GstWasapi2Client * self, guint32 * n_frames)
-{
- HRESULT hr;
- UINT32 n_frames_padding;
- IAudioClient3 *audio_client = self->audio_client;
-
- *n_frames = 0;
-
- if (!audio_client) {
- GST_WARNING_OBJECT (self, "IAudioClient3 wasn't configured");
- return E_FAIL;
- }
-
- /* Frames the card hasn't rendered yet */
- hr = audio_client->GetCurrentPadding (&n_frames_padding);
- if (!gst_wasapi2_result (hr))
- return hr;
-
- GST_LOG_OBJECT (self, "%d unread frames (padding)", n_frames_padding);
-
- /* We can write out these many frames */
- *n_frames = self->buffer_frame_count - n_frames_padding;
-
- return S_OK;
-}
-
-HRESULT
-gst_wasapi2_client_start (GstWasapi2Client * client)
-{
- HRESULT hr;
- IAudioClient3 *audio_client;
- WAVEFORMATEX *mix_format;
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
-
- if (client->running) {
- GST_WARNING_OBJECT (client, "IAudioClient3 is running already");
- return S_OK;
- }
-
- audio_client = client->audio_client;
- mix_format = client->mix_format;
-
- if (!audio_client) {
- GST_ERROR_OBJECT (client, "IAudioClient object wasn't configured");
- return E_FAIL;
- }
-
- if (!mix_format) {
- GST_ERROR_OBJECT (client, "Unknown MixFormat");
- return E_FAIL;
- }
-
- if (client->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE &&
- !client->audio_capture_client) {
- GST_ERROR_OBJECT (client, "IAudioCaptureClient wasn't configured");
- return E_FAIL;
- }
-
- if (client->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER &&
- !client->audio_render_client) {
- GST_ERROR_OBJECT (client, "IAudioRenderClient wasn't configured");
- return E_FAIL;
- }
-
- ResetEvent (client->cancellable);
-
- /* To avoid start-up glitches, before starting the streaming, we fill the
- * buffer with silence as recommended by the documentation:
- * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370879%28v=vs.85%29.aspx */
- if (client->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER) {
- IAudioRenderClient *render_client = client->audio_render_client;
- guint32 n_frames, len;
- BYTE *dst = NULL;
-
- hr = gst_wasapi2_client_get_can_frames (client, &n_frames);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client,
- "should have more than %i frames to write", n_frames);
- return hr;
- }
-
- len = n_frames * mix_format->nBlockAlign;
-
- hr = render_client->GetBuffer (n_frames, &dst);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client, "Couldn't get buffer");
- return hr;
- }
-
- GST_DEBUG_OBJECT (client, "pre-wrote %i bytes of silence", len);
-
- hr = render_client->ReleaseBuffer (n_frames, AUDCLNT_BUFFERFLAGS_SILENT);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client, "Couldn't release buffer");
- return hr;
- }
- }
-
- hr = audio_client->Start ();
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client, "Couldn't start audio client");
- return hr;
- }
-
- client->running = TRUE;
- gst_adapter_clear (client->adapter);
-
- return S_OK;
-}
-
-HRESULT
-gst_wasapi2_client_stop (GstWasapi2Client * client)
-{
- HRESULT hr;
- IAudioClient3 *audio_client;
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
-
- audio_client = client->audio_client;
-
- if (!client->running) {
- GST_DEBUG_OBJECT (client, "We are not running now");
- return S_OK;
- }
-
- if (!client->audio_client) {
- GST_ERROR_OBJECT (client, "IAudioClient3 object wasn't configured");
- return E_FAIL;
- }
-
- client->running = FALSE;
- SetEvent (client->cancellable);
-
- hr = audio_client->Stop ();
- if (!gst_wasapi2_result (hr))
- return hr;
-
- /* reset state for reuse case */
- return audio_client->Reset ();
-}
-
-HRESULT
-gst_wasapi2_client_read (GstWasapi2Client * client, gpointer data, guint length,
- guint * read_length)
-{
- IAudioCaptureClient *capture_client;
- WAVEFORMATEX *mix_format;
- HRESULT hr;
- BYTE *from = NULL;
- guint wanted = length;
- guint bpf;
- DWORD flags;
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
- g_return_val_if_fail (client->audio_capture_client != NULL, E_INVALIDARG);
- g_return_val_if_fail (client->mix_format != NULL, E_INVALIDARG);
- g_return_val_if_fail (read_length != NULL, E_INVALIDARG);
-
- *read_length = 0;
-
- capture_client = client->audio_capture_client;
- mix_format = client->mix_format;
-
- if (!client->running) {
- GST_ERROR_OBJECT (client, "client is not running now");
- return E_FAIL;
- }
-
- /* If we've accumulated enough data, return it immediately */
- if (gst_adapter_available (client->adapter) >= wanted) {
- memcpy (data, gst_adapter_map (client->adapter, wanted), wanted);
- gst_adapter_flush (client->adapter, wanted);
- GST_DEBUG_OBJECT (client, "Adapter has enough data, returning %i", wanted);
-
- *read_length = wanted;
-
- return S_OK;
- }
-
- bpf = mix_format->nBlockAlign;
-
- while (wanted > 0) {
- DWORD dwWaitResult;
- guint got_frames, avail_frames, n_frames, want_frames, read_len;
- HANDLE event_handle[2];
-
- event_handle[0] = client->event_handle;
- event_handle[1] = client->cancellable;
-
- /* Wait for data to become available */
- dwWaitResult = WaitForMultipleObjects (2, event_handle, FALSE, INFINITE);
- if (dwWaitResult != WAIT_OBJECT_0 && dwWaitResult != WAIT_OBJECT_0 + 1) {
- GST_ERROR_OBJECT (client, "Error waiting for event handle: %x",
- (guint) dwWaitResult);
- return E_FAIL;
- }
-
- if (!client->running) {
- GST_DEBUG_OBJECT (client, "Cancelled");
- return S_OK;
- }
-
- hr = capture_client->GetBuffer (&from, &got_frames, &flags, nullptr,
- nullptr);
- if (!gst_wasapi2_result (hr)) {
- if (hr == AUDCLNT_S_BUFFER_EMPTY) {
- GST_INFO_OBJECT (client, "Client buffer is empty, retry");
- return S_OK;
- }
-
- GST_ERROR_OBJECT (client, "Couldn't get buffer from capture client");
- return hr;
- }
-
- if (got_frames == 0) {
- GST_DEBUG_OBJECT (client, "No buffer to read");
- capture_client->ReleaseBuffer (got_frames);
- return S_OK;
- }
-
- if (G_UNLIKELY (flags != 0)) {
- /* https://docs.microsoft.com/en-us/windows/win32/api/audioclient/ne-audioclient-_audclnt_bufferflags */
- if (flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY)
- GST_DEBUG_OBJECT (client, "WASAPI reported discontinuity (glitch?)");
- if (flags & AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR)
- GST_DEBUG_OBJECT (client, "WASAPI reported a timestamp error");
- }
-
- /* Copy all the frames we got into the adapter, and then extract at most
- * @wanted size of frames from it. This helps when ::GetBuffer returns more
- * data than we can handle right now. */
- {
- GstBuffer *tmp = gst_buffer_new_allocate (NULL, got_frames * bpf, NULL);
- /* If flags has AUDCLNT_BUFFERFLAGS_SILENT, we will ignore the actual
- * data and write out silence, see:
- * https://docs.microsoft.com/en-us/windows/win32/api/audioclient/ne-audioclient-_audclnt_bufferflags */
- if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
- memset (from, 0, got_frames * bpf);
- gst_buffer_fill (tmp, 0, from, got_frames * bpf);
- gst_adapter_push (client->adapter, tmp);
- }
-
- /* Release all captured buffers; we copied them above */
- hr = capture_client->ReleaseBuffer (got_frames);
- from = NULL;
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client, "Failed to release buffer");
- return hr;
- }
-
- want_frames = wanted / bpf;
- avail_frames = gst_adapter_available (client->adapter) / bpf;
-
- /* Only copy data that will fit into the allocated buffer of size @length */
- n_frames = MIN (avail_frames, want_frames);
- read_len = n_frames * bpf;
-
- if (read_len == 0) {
- GST_WARNING_OBJECT (client, "No data to read");
- return S_OK;
- }
-
- GST_LOG_OBJECT (client, "frames captured: %d (%d bytes), "
- "can read: %d (%d bytes), will read: %d (%d bytes), "
- "adapter has: %d (%d bytes)", got_frames, got_frames * bpf, want_frames,
- wanted, n_frames, read_len, avail_frames, avail_frames * bpf);
-
- memcpy (data, gst_adapter_map (client->adapter, read_len), read_len);
- gst_adapter_flush (client->adapter, read_len);
- wanted -= read_len;
- }
-
- *read_length = length;
-
- return S_OK;
-}
-
-HRESULT
-gst_wasapi2_client_write (GstWasapi2Client * client, gpointer data,
- guint length, guint * write_length)
-{
- IAudioRenderClient *render_client;
- WAVEFORMATEX *mix_format;
- HRESULT hr;
- BYTE *dst = nullptr;
- DWORD dwWaitResult;
- guint can_frames, have_frames, n_frames, write_len = 0;
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
- g_return_val_if_fail (client->audio_render_client != NULL, E_INVALIDARG);
- g_return_val_if_fail (client->mix_format != NULL, E_INVALIDARG);
- g_return_val_if_fail (write_length != NULL, E_INVALIDARG);
-
- *write_length = 0;
-
- if (!client->running) {
- GST_WARNING_OBJECT (client, "client is not running now");
- return -1;
- }
-
- render_client = client->audio_render_client;
- mix_format = client->mix_format;
-
- /* We have N frames to be written out */
- have_frames = length / (mix_format->nBlockAlign);
-
- /* In shared mode we can write parts of the buffer, so only wait
- * in case we can't write anything */
- hr = gst_wasapi2_client_get_can_frames (client, &can_frames);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client, "Error getting frames to write to");
- return hr;
- }
-
- if (can_frames == 0) {
- HANDLE event_handle[2];
-
- event_handle[0] = client->event_handle;
- event_handle[1] = client->cancellable;
-
- dwWaitResult = WaitForMultipleObjects (2, event_handle, FALSE, INFINITE);
- if (dwWaitResult != WAIT_OBJECT_0 && dwWaitResult != WAIT_OBJECT_0 + 1) {
- GST_ERROR_OBJECT (client, "Error waiting for event handle: %x",
- (guint) dwWaitResult);
- return E_FAIL;
- }
-
- if (!client->running) {
- GST_DEBUG_OBJECT (client, "Cancelled");
- return S_OK;
- }
-
- hr = gst_wasapi2_client_get_can_frames (client, &can_frames);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client, "Error getting frames to write to");
- return hr;
- }
- }
-
- /* We will write out these many frames, and this much length */
- n_frames = MIN (can_frames, have_frames);
- write_len = n_frames * mix_format->nBlockAlign;
-
- GST_LOG_OBJECT (client, "total: %d, have_frames: %d (%d bytes), "
- "can_frames: %d, will write: %d (%d bytes)", client->buffer_frame_count,
- have_frames, length, can_frames, n_frames, write_len);
-
- hr = render_client->GetBuffer (n_frames, &dst);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client, "Couldn't get buffer from client");
- return hr;
- }
-
- memcpy (dst, data, write_len);
- hr = render_client->ReleaseBuffer (n_frames, 0);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (client, "Failed to release buffer");
- return hr;
- }
-
- *write_length = write_len;
-
- return S_OK;
-}
-
-HRESULT
-gst_wasapi2_client_delay (GstWasapi2Client * client, guint32 * delay)
-{
- IAudioClient3 *audio_client;
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
- g_return_val_if_fail (delay != nullptr, E_INVALIDARG);
-
- *delay = 0;
-
- audio_client = client->audio_client;
- if (!audio_client) {
- GST_WARNING_OBJECT (client, "IAudioClient3 wasn't configured");
- return E_FAIL;
- }
-
- return audio_client->GetCurrentPadding (delay);
-}
-
-HRESULT
-gst_wasapi2_client_set_mute (GstWasapi2Client * client, gboolean mute)
-{
- HRESULT hr;
- ISimpleAudioVolume *audio_volume;
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
-
- audio_volume = client->audio_volume;
-
- if (!audio_volume) {
- GST_WARNING_OBJECT (client, "ISimpleAudioVolume object wasn't configured");
- return E_FAIL;
- }
-
- hr = audio_volume->SetMute (mute, nullptr);
- GST_DEBUG_OBJECT (client, "Set mute %s, hr: 0x%x",
- mute ? "enabled" : "disabled", (gint) hr);
-
- return hr;
-}
-
-HRESULT
-gst_wasapi2_client_get_mute (GstWasapi2Client * client, gboolean * mute)
-{
- HRESULT hr;
- ISimpleAudioVolume *audio_volume;
- BOOL current_mute = FALSE;
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
- g_return_val_if_fail (mute != NULL, E_INVALIDARG);
-
- audio_volume = client->audio_volume;
-
- if (!audio_volume) {
- GST_WARNING_OBJECT (client, "ISimpleAudioVolume object wasn't configured");
- return E_FAIL;
- }
-
- hr = audio_volume->GetMute (&current_mute);
- if (!gst_wasapi2_result (hr))
- return hr;
-
- *mute = (gboolean) current_mute;
-
- return S_OK;
-}
-
-HRESULT
-gst_wasapi2_client_set_volume (GstWasapi2Client * client, gfloat volume)
-{
- HRESULT hr;
- ISimpleAudioVolume *audio_volume;
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
- g_return_val_if_fail (volume >= 0 && volume <= 1.0, E_INVALIDARG);
-
- audio_volume = client->audio_volume;
-
- if (!audio_volume) {
- GST_WARNING_OBJECT (client, "ISimpleAudioVolume object wasn't configured");
- return E_FAIL;
- }
-
- hr = audio_volume->SetMasterVolume (volume, nullptr);
- GST_DEBUG_OBJECT (client, "Set volume %.2f hr: 0x%x", volume, (gint) hr);
-
- return hr;
-}
-
-HRESULT
-gst_wasapi2_client_get_volume (GstWasapi2Client * client, gfloat * volume)
-{
- HRESULT hr;
- ISimpleAudioVolume *audio_volume;
- float current_volume = FALSE;
-
- g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), E_INVALIDARG);
- g_return_val_if_fail (volume != NULL, E_INVALIDARG);
-
- audio_volume = client->audio_volume;
-
- if (!audio_volume) {
- GST_WARNING_OBJECT (client, "ISimpleAudioVolume object wasn't configured");
- return E_FAIL;
- }
-
- hr = audio_volume->GetMasterVolume (&current_volume);
- if (!gst_wasapi2_result (hr))
- return hr;
-
- *volume = current_volume;
-
- return S_OK;
-}
-
gboolean
gst_wasapi2_client_ensure_activation (GstWasapi2Client * client)
{
@@ -1980,8 +917,7 @@ find_dispatcher (ICoreDispatcher ** dispatcher)
GstWasapi2Client *
gst_wasapi2_client_new (GstWasapi2ClientDeviceClass device_class,
- gboolean low_latency, gint device_index, const gchar * device_id,
- gpointer dispatcher)
+ gint device_index, const gchar * device_id, gpointer dispatcher)
{
GstWasapi2Client *self;
/* *INDENT-OFF* */
@@ -2007,9 +943,8 @@ gst_wasapi2_client_new (GstWasapi2ClientDeviceClass device_class,
}
self = (GstWasapi2Client *) g_object_new (GST_TYPE_WASAPI2_CLIENT,
- "device-class", device_class, "low-latency", low_latency,
- "device-index", device_index, "device", device_id,
- "dispatcher", dispatcher, NULL);
+ "device-class", device_class, "device-index", device_index,
+ "device", device_id, "dispatcher", dispatcher, nullptr);
/* Reset explicitly to ensure that it happens before
* RoInitializeWrapper dtor is called */
@@ -2017,10 +952,18 @@ gst_wasapi2_client_new (GstWasapi2ClientDeviceClass device_class,
if (self->activate_state == GST_WASAPI2_CLIENT_ACTIVATE_FAILED) {
gst_object_unref (self);
- return NULL;
+ return nullptr;
}
gst_object_ref_sink (self);
return self;
}
+
+IAudioClient *
+gst_wasapi2_client_get_handle (GstWasapi2Client * client)
+{
+ g_return_val_if_fail (GST_IS_WASAPI2_CLIENT (client), nullptr);
+
+ return client->audio_client;
+}
diff --git a/sys/wasapi2/gstwasapi2client.h b/sys/wasapi2/gstwasapi2client.h
index 57b281101..6c8c78e21 100644
--- a/sys/wasapi2/gstwasapi2client.h
+++ b/sys/wasapi2/gstwasapi2client.h
@@ -39,50 +39,16 @@ GType gst_wasapi2_client_device_class_get_type (void);
G_DECLARE_FINAL_TYPE (GstWasapi2Client,
gst_wasapi2_client, GST, WASAPI2_CLIENT, GstObject);
-GstCaps * gst_wasapi2_client_get_caps (GstWasapi2Client * client);
-
-HRESULT gst_wasapi2_client_open (GstWasapi2Client * client,
- GstAudioRingBufferSpec * spec,
- GstAudioRingBuffer * buf);
-
-HRESULT gst_wasapi2_client_start (GstWasapi2Client * client);
-
-HRESULT gst_wasapi2_client_stop (GstWasapi2Client * client);
-
-HRESULT gst_wasapi2_client_read (GstWasapi2Client * client,
- gpointer data,
- guint length,
- guint * read_length);
-
-HRESULT gst_wasapi2_client_write (GstWasapi2Client * client,
- gpointer data,
- guint length,
- guint * write_length);
-
-HRESULT gst_wasapi2_client_delay (GstWasapi2Client * client,
- guint32 * delay);
-
-HRESULT gst_wasapi2_client_set_mute (GstWasapi2Client * client,
- gboolean mute);
-
-HRESULT gst_wasapi2_client_get_mute (GstWasapi2Client * client,
- gboolean * mute);
-
-HRESULT gst_wasapi2_client_set_volume (GstWasapi2Client * client,
- gfloat volume);
-
-HRESULT gst_wasapi2_client_get_volume (GstWasapi2Client * client,
- gfloat * volume);
-
-gboolean gst_wasapi2_client_ensure_activation (GstWasapi2Client * client);
-
GstWasapi2Client * gst_wasapi2_client_new (GstWasapi2ClientDeviceClass device_class,
- gboolean low_latency,
gint device_index,
const gchar * device_id,
gpointer dispatcher);
-G_DEFINE_AUTOPTR_CLEANUP_FUNC (GstWasapi2Client, gst_object_unref)
+gboolean gst_wasapi2_client_ensure_activation (GstWasapi2Client * client);
+
+IAudioClient * gst_wasapi2_client_get_handle (GstWasapi2Client * client);
+
+GstCaps * gst_wasapi2_client_get_caps (GstWasapi2Client * client);
G_END_DECLS
diff --git a/sys/wasapi2/gstwasapi2device.c b/sys/wasapi2/gstwasapi2device.c
index 6a36fcecd..4fd9abfaf 100644
--- a/sys/wasapi2/gstwasapi2device.c
+++ b/sys/wasapi2/gstwasapi2device.c
@@ -183,7 +183,7 @@ gst_wasapi2_device_provider_probe_internal (GstWasapi2DeviceProvider * self,
gchar *device_id = NULL;
gchar *device_name = NULL;
- client = gst_wasapi2_client_new (client_class, FALSE, i, NULL, NULL);
+ client = gst_wasapi2_client_new (client_class, i, NULL, NULL);
if (!client)
return;
diff --git a/sys/wasapi2/gstwasapi2ringbuffer.cpp b/sys/wasapi2/gstwasapi2ringbuffer.cpp
new file mode 100644
index 000000000..5a85458b3
--- /dev/null
+++ b/sys/wasapi2/gstwasapi2ringbuffer.cpp
@@ -0,0 +1,1015 @@
+/* GStreamer
+ * Copyright (C) 2021 Seungha Yang <seungha@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.
+ */
+
+#include "gstwasapi2ringbuffer.h"
+#include <string.h>
+#include <mfapi.h>
+#include <wrl.h>
+
+GST_DEBUG_CATEGORY_STATIC (gst_wasapi2_ring_buffer_debug);
+#define GST_CAT_DEFAULT gst_wasapi2_ring_buffer_debug
+
+static HRESULT gst_wasapi2_ring_buffer_io_callback (GstWasapi2RingBuffer * buf);
+
+/* *INDENT-OFF* */
+using namespace Microsoft::WRL;
+
+class GstWasapiAsyncCallback : public IMFAsyncCallback
+{
+public:
+ GstWasapiAsyncCallback(GstWasapi2RingBuffer *listener, DWORD queue_id)
+ : ref_count_(1)
+ , queue_id_(queue_id)
+ {
+ g_weak_ref_init (&listener_, listener);
+ }
+
+ virtual ~GstWasapiAsyncCallback ()
+ {
+ g_weak_ref_set (&listener_, nullptr);
+ }
+
+ /* IUnknown */
+ STDMETHODIMP_ (ULONG)
+ AddRef (void)
+ {
+ GST_TRACE ("%p, %d", this, ref_count_);
+ return InterlockedIncrement (&ref_count_);
+ }
+ STDMETHODIMP_ (ULONG)
+ Release (void)
+ {
+ ULONG ref_count;
+
+ GST_TRACE ("%p, %d", this, ref_count_);
+ ref_count = InterlockedDecrement (&ref_count_);
+
+ if (ref_count == 0) {
+ GST_TRACE ("Delete instance %p", this);
+ delete this;
+ }
+
+ return ref_count;
+ }
+
+ STDMETHODIMP
+ QueryInterface (REFIID riid, void ** object)
+ {
+ if (!object)
+ return E_POINTER;
+
+ if (riid == IID_IUnknown) {
+ GST_TRACE ("query IUnknown interface %p", this);
+ *object = static_cast<IUnknown *> (static_cast<GstWasapiAsyncCallback *> (this));
+ } else if (riid == __uuidof (IMFAsyncCallback)) {
+ GST_TRACE ("query IUnknown interface %p", this);
+ *object = static_cast<IUnknown *> (static_cast<GstWasapiAsyncCallback *> (this));
+ } else {
+ *object = nullptr;
+ return E_NOINTERFACE;
+ }
+
+ AddRef ();
+
+ return S_OK;
+ }
+
+ /* IMFAsyncCallback */
+ STDMETHODIMP
+ GetParameters(DWORD * pdwFlags, DWORD * pdwQueue)
+ {
+ *pdwFlags = 0;
+ *pdwQueue = queue_id_;
+
+ return S_OK;
+ }
+
+ STDMETHODIMP
+ Invoke(IMFAsyncResult * pAsyncResult)
+ {
+ GstWasapi2RingBuffer *ringbuffer;
+ HRESULT hr;
+
+ ringbuffer = (GstWasapi2RingBuffer *) g_weak_ref_get (&listener_);
+ if (!ringbuffer) {
+ GST_WARNING ("Listener was removed");
+ return S_OK;
+ }
+
+ hr = gst_wasapi2_ring_buffer_io_callback (ringbuffer);
+ gst_object_unref (ringbuffer);
+
+ return hr;
+ }
+
+private:
+ ULONG ref_count_;
+ DWORD queue_id_;
+ GWeakRef listener_;
+};
+/* *INDENT-ON* */
+
+struct _GstWasapi2RingBuffer
+{
+ GstAudioRingBuffer parent;
+
+ GstWasapi2ClientDeviceClass device_class;
+ gchar *device_id;
+ gboolean low_latency;
+ gboolean mute;
+ gdouble volume;
+ gpointer dispatcher;
+
+ GstWasapi2Client *client;
+ IAudioCaptureClient *capture_client;
+ IAudioRenderClient *render_client;
+ ISimpleAudioVolume *volume_object;
+
+ GstWasapiAsyncCallback *callback_object;
+ IMFAsyncResult *callback_result;
+ MFWORKITEM_KEY callback_key;
+ HANDLE event_handle;
+
+ guint64 expected_position;
+ gboolean is_first;
+ gboolean running;
+ UINT32 buffer_size;
+
+ gint segoffset;
+ guint64 write_frame_offset;
+
+ GMutex volume_lock;
+ gboolean mute_changed;
+ gboolean volume_changed;
+};
+
+static void gst_wasapi2_ring_buffer_constructed (GObject * object);
+static void gst_wasapi2_ring_buffer_dispose (GObject * object);
+static void gst_wasapi2_ring_buffer_finalize (GObject * object);
+
+static gboolean gst_wasapi2_ring_buffer_open_device (GstAudioRingBuffer * buf);
+static gboolean gst_wasapi2_ring_buffer_close_device (GstAudioRingBuffer * buf);
+static gboolean gst_wasapi2_ring_buffer_acquire (GstAudioRingBuffer * buf,
+ GstAudioRingBufferSpec * spec);
+static gboolean gst_wasapi2_ring_buffer_release (GstAudioRingBuffer * buf);
+static gboolean gst_wasapi2_ring_buffer_start (GstAudioRingBuffer * buf);
+static gboolean gst_wasapi2_ring_buffer_stop (GstAudioRingBuffer * buf);
+static guint gst_wasapi2_ring_buffer_delay (GstAudioRingBuffer * buf);
+
+#define gst_wasapi2_ring_buffer_parent_class parent_class
+G_DEFINE_TYPE (GstWasapi2RingBuffer, gst_wasapi2_ring_buffer,
+ GST_TYPE_AUDIO_RING_BUFFER);
+
+static void
+gst_wasapi2_ring_buffer_class_init (GstWasapi2RingBufferClass * klass)
+{
+ GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+ GstAudioRingBufferClass *ring_buffer_class =
+ GST_AUDIO_RING_BUFFER_CLASS (klass);
+
+ gobject_class->constructed = gst_wasapi2_ring_buffer_constructed;
+ gobject_class->dispose = gst_wasapi2_ring_buffer_dispose;
+ gobject_class->finalize = gst_wasapi2_ring_buffer_finalize;
+
+ ring_buffer_class->open_device =
+ GST_DEBUG_FUNCPTR (gst_wasapi2_ring_buffer_open_device);
+ ring_buffer_class->close_device =
+ GST_DEBUG_FUNCPTR (gst_wasapi2_ring_buffer_close_device);
+ ring_buffer_class->acquire =
+ GST_DEBUG_FUNCPTR (gst_wasapi2_ring_buffer_acquire);
+ ring_buffer_class->release =
+ GST_DEBUG_FUNCPTR (gst_wasapi2_ring_buffer_release);
+ ring_buffer_class->start = GST_DEBUG_FUNCPTR (gst_wasapi2_ring_buffer_start);
+ ring_buffer_class->resume = GST_DEBUG_FUNCPTR (gst_wasapi2_ring_buffer_start);
+ ring_buffer_class->stop = GST_DEBUG_FUNCPTR (gst_wasapi2_ring_buffer_stop);
+ ring_buffer_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi2_ring_buffer_delay);
+
+ GST_DEBUG_CATEGORY_INIT (gst_wasapi2_ring_buffer_debug,
+ "wasapi2ringbuffer", 0, "wasapi2ringbuffer");
+}
+
+static void
+gst_wasapi2_ring_buffer_init (GstWasapi2RingBuffer * self)
+{
+ self->volume = 1.0f;
+ self->mute = FALSE;
+
+ self->event_handle = CreateEvent (nullptr, FALSE, FALSE, nullptr);
+ g_mutex_init (&self->volume_lock);
+}
+
+static void
+gst_wasapi2_ring_buffer_constructed (GObject * object)
+{
+ GstWasapi2RingBuffer *self = GST_WASAPI2_RING_BUFFER (object);
+ HRESULT hr;
+ DWORD task_id = 0;
+ DWORD queue_id = 0;
+
+ hr = MFLockSharedWorkQueue (L"Pro Audio", 0, &task_id, &queue_id);
+ if (!gst_wasapi2_result (hr)) {
+ GST_WARNING_OBJECT (self, "Failed to get work queue id");
+ goto out;
+ }
+
+ self->callback_object = new GstWasapiAsyncCallback (self, queue_id);
+ hr = MFCreateAsyncResult (nullptr, self->callback_object, nullptr,
+ &self->callback_result);
+ if (!gst_wasapi2_result (hr)) {
+ GST_WARNING_OBJECT (self, "Failed to create IAsyncResult");
+ GST_WASAPI2_CLEAR_COM (self->callback_object);
+ }
+
+out:
+ G_OBJECT_CLASS (parent_class)->constructed (object);
+}
+
+static void
+gst_wasapi2_ring_buffer_dispose (GObject * object)
+{
+ GstWasapi2RingBuffer *self = GST_WASAPI2_RING_BUFFER (object);
+
+ gst_clear_object (&self->client);
+ GST_WASAPI2_CLEAR_COM (self->callback_result);
+ GST_WASAPI2_CLEAR_COM (self->callback_object);
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_wasapi2_ring_buffer_finalize (GObject * object)
+{
+ GstWasapi2RingBuffer *self = GST_WASAPI2_RING_BUFFER (object);
+
+ g_free (self->device_id);
+ CloseHandle (self->event_handle);
+ g_mutex_clear (&self->volume_lock);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_wasapi2_ring_buffer_open_device (GstAudioRingBuffer * buf)
+{
+ GstWasapi2RingBuffer *self = GST_WASAPI2_RING_BUFFER (buf);
+
+ GST_DEBUG_OBJECT (self, "Open");
+
+ self->client = gst_wasapi2_client_new (self->device_class,
+ -1, self->device_id, self->dispatcher);
+ if (!self->client) {
+ GST_ERROR_OBJECT (self, "Failed to open device");
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+static gboolean
+gst_wasapi2_ring_buffer_close_device (GstAudioRingBuffer * buf)
+{
+ GstWasapi2RingBuffer *self = GST_WASAPI2_RING_BUFFER (buf);
+
+ GST_DEBUG_OBJECT (self, "Close");
+
+ GST_WASAPI2_CLEAR_COM (self->capture_client);
+ GST_WASAPI2_CLEAR_COM (self->render_client);
+
+ g_mutex_lock (&self->volume_lock);
+ if (self->volume_object)
+ self->volume_object->SetMute (FALSE, nullptr);
+ GST_WASAPI2_CLEAR_COM (self->volume_object);
+ g_mutex_unlock (&self->volume_lock);
+
+ gst_clear_object (&self->client);
+
+ return TRUE;
+}
+
+static HRESULT
+gst_wasapi2_ring_buffer_read (GstWasapi2RingBuffer * self)
+{
+ GstAudioRingBuffer *ringbuffer = GST_AUDIO_RING_BUFFER_CAST (self);
+ BYTE *data = nullptr;
+ UINT32 to_read = 0;
+ guint32 to_read_bytes;
+ DWORD flags = 0;
+ HRESULT hr;
+ guint64 position;
+ GstAudioInfo *info = &ringbuffer->spec.info;
+ IAudioCaptureClient *capture_client = self->capture_client;
+ guint gap_size = 0;
+ guint offset = 0;
+ gint segment;
+ guint8 *readptr;
+ gint len;
+
+ if (!capture_client) {
+ GST_ERROR_OBJECT (self, "IAudioCaptureClient is not available");
+ return E_FAIL;
+ }
+
+ hr = capture_client->GetBuffer (&data, &to_read, &flags, &position, nullptr);
+ if (hr == AUDCLNT_S_BUFFER_EMPTY || to_read == 0) {
+ GST_LOG_OBJECT (self, "Empty buffer");
+ to_read = 0;
+ goto out;
+ }
+
+ to_read_bytes = to_read * GST_AUDIO_INFO_BPF (info);
+
+ GST_TRACE_OBJECT (self, "Reading at %d frames offset %" G_GUINT64_FORMAT
+ ", expected position %" G_GUINT64_FORMAT, to_read, position,
+ self->expected_position);
+
+ if (self->is_first) {
+ self->expected_position = position + to_read;
+ self->is_first = FALSE;
+ } else {
+ if (position > self->expected_position) {
+ guint gap_frames;
+
+ gap_frames = (guint) (position - self->expected_position);
+ GST_WARNING_OBJECT (self, "Found %u frames gap", gap_frames);
+ gap_size = gap_frames * GST_AUDIO_INFO_BPF (info);
+ }
+
+ self->expected_position = position + to_read;
+ }
+
+ /* Fill gap data if any */
+ while (gap_size > 0) {
+ if (!gst_audio_ring_buffer_prepare_read (ringbuffer,
+ &segment, &readptr, &len)) {
+ GST_INFO_OBJECT (self, "No segment available");
+ goto out;
+ }
+
+ g_assert (self->segoffset >= 0);
+
+ len -= self->segoffset;
+ if (len > gap_size)
+ len = gap_size;
+
+ gst_audio_format_info_fill_silence (ringbuffer->spec.info.finfo,
+ readptr + self->segoffset, len);
+
+ self->segoffset += len;
+ gap_size -= len;
+
+ if (self->segoffset == ringbuffer->spec.segsize) {
+ gst_audio_ring_buffer_advance (ringbuffer, 1);
+ self->segoffset = 0;
+ }
+ }
+
+ while (to_read_bytes) {
+ if (!gst_audio_ring_buffer_prepare_read (ringbuffer,
+ &segment, &readptr, &len)) {
+ GST_INFO_OBJECT (self, "No segment available");
+ goto out;
+ }
+
+ len -= self->segoffset;
+ if (len > to_read_bytes)
+ len = to_read_bytes;
+
+ memcpy (readptr + self->segoffset, data + offset, len);
+
+ self->segoffset += len;
+ offset += len;
+ to_read_bytes -= len;
+
+ if (self->segoffset == ringbuffer->spec.segsize) {
+ gst_audio_ring_buffer_advance (ringbuffer, 1);
+ self->segoffset = 0;
+ }
+ }
+
+out:
+ hr = capture_client->ReleaseBuffer (to_read);
+ /* For debugging */
+ gst_wasapi2_result (hr);
+
+ return S_OK;
+}
+
+static HRESULT
+gst_wasapi2_ring_buffer_write (GstWasapi2RingBuffer * self, gboolean preroll)
+{
+ GstAudioRingBuffer *ringbuffer = GST_AUDIO_RING_BUFFER_CAST (self);
+ HRESULT hr;
+ IAudioClient *client_handle;
+ IAudioRenderClient *render_client;
+ guint32 padding_frames = 0;
+ guint32 can_write;
+ guint32 can_write_bytes;
+ gint segment;
+ guint8 *readptr;
+ gint len;
+ BYTE *data = nullptr;
+
+ client_handle = gst_wasapi2_client_get_handle (self->client);
+ if (!client_handle) {
+ GST_ERROR_OBJECT (self, "IAudioClient is not available");
+ return E_FAIL;
+ }
+
+ render_client = self->render_client;
+ if (!render_client) {
+ GST_ERROR_OBJECT (self, "IAudioRenderClient is not available");
+ return E_FAIL;
+ }
+
+ hr = client_handle->GetCurrentPadding (&padding_frames);
+ if (!gst_wasapi2_result (hr))
+ return hr;
+
+ if (padding_frames >= self->buffer_size) {
+ GST_INFO_OBJECT (self,
+ "Padding size %d is larger than or equal to buffer size %d",
+ padding_frames, self->buffer_size);
+ return S_OK;
+ }
+
+ can_write = self->buffer_size - padding_frames;
+ can_write_bytes = can_write * GST_AUDIO_INFO_BPF (&ringbuffer->spec.info);
+ if (preroll) {
+ GST_INFO_OBJECT (self, "Pre-fill %d frames with silence", can_write);
+
+ hr = render_client->GetBuffer (can_write, &data);
+ if (!gst_wasapi2_result (hr))
+ return hr;
+
+ hr = render_client->ReleaseBuffer (can_write, AUDCLNT_BUFFERFLAGS_SILENT);
+ return gst_wasapi2_result (hr);
+ }
+
+ GST_TRACE_OBJECT (self, "Writing %d frames offset at %" G_GUINT64_FORMAT,
+ can_write, self->write_frame_offset);
+ self->write_frame_offset += can_write;
+
+ while (can_write_bytes > 0) {
+ if (!gst_audio_ring_buffer_prepare_read (ringbuffer,
+ &segment, &readptr, &len)) {
+ GST_INFO_OBJECT (self, "No segment available");
+ return S_OK;
+ }
+
+ len -= self->segoffset;
+
+ if (len > can_write_bytes)
+ len = can_write_bytes;
+
+ can_write = len / GST_AUDIO_INFO_BPF (&ringbuffer->spec.info);
+ if (can_write == 0)
+ break;
+
+ hr = render_client->GetBuffer (can_write, &data);
+ if (!gst_wasapi2_result (hr))
+ return hr;
+
+ memcpy (data, readptr + self->segoffset, len);
+ hr = render_client->ReleaseBuffer (can_write, 0);
+
+ self->segoffset += len;
+ can_write_bytes -= len;
+
+ if (self->segoffset == ringbuffer->spec.segsize) {
+ gst_audio_ring_buffer_clear (ringbuffer, segment);
+ gst_audio_ring_buffer_advance (ringbuffer, 1);
+ self->segoffset = 0;
+ }
+
+ if (!gst_wasapi2_result (hr)) {
+ GST_WARNING_OBJECT (self, "Failed to release buffer");
+ break;
+ }
+ }
+
+ return S_OK;
+}
+
+static HRESULT
+gst_wasapi2_ring_buffer_io_callback (GstWasapi2RingBuffer * self)
+{
+ HRESULT hr = E_FAIL;
+
+ g_return_val_if_fail (GST_IS_WASAPI2_RING_BUFFER (self), E_FAIL);
+
+ if (!self->running) {
+ GST_INFO_OBJECT (self, "We are not running now");
+ return S_OK;
+ }
+
+ switch (self->device_class) {
+ case GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE:
+ hr = gst_wasapi2_ring_buffer_read (self);
+ break;
+ case GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER:
+ hr = gst_wasapi2_ring_buffer_write (self, FALSE);
+ break;
+ default:
+ g_assert_not_reached ();
+ break;
+ }
+
+ if (self->running) {
+ if (gst_wasapi2_result (hr)) {
+ hr = MFPutWaitingWorkItem (self->event_handle, 0, self->callback_result,
+ &self->callback_key);
+
+ if (!gst_wasapi2_result (hr)) {
+ GstElement *parent =
+ (GstElement *) gst_object_get_parent (GST_OBJECT_CAST (self));
+
+ GST_ERROR_OBJECT (self, "Failed to put item");
+ if (parent) {
+ GST_ELEMENT_ERROR (parent, RESOURCE, FAILED,
+ (nullptr), ("Failed to schedule next I/O"));
+ gst_object_unref (parent);
+ }
+ }
+ }
+ } else {
+ GST_INFO_OBJECT (self, "We are not running now");
+ return S_OK;
+ }
+
+ return hr;
+}
+
+static HRESULT
+gst_wasapi2_ring_buffer_initialize_audio_client3 (GstWasapi2RingBuffer * self,
+ IAudioClient * client_handle, WAVEFORMATEX * mix_format, guint * period)
+{
+ HRESULT hr = S_OK;
+ UINT32 default_period, fundamental_period, min_period, max_period;
+ /* AUDCLNT_STREAMFLAGS_NOPERSIST is not allowed for
+ * InitializeSharedAudioStream */
+ DWORD stream_flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
+ ComPtr < IAudioClient3 > audio_client;
+
+ hr = client_handle->QueryInterface (IID_PPV_ARGS (&audio_client));
+ if (!gst_wasapi2_result (hr)) {
+ GST_INFO_OBJECT (self, "IAudioClient3 interface is unavailable");
+ return hr;
+ }
+
+ hr = audio_client->GetSharedModeEnginePeriod (mix_format,
+ &default_period, &fundamental_period, &min_period, &max_period);
+ if (!gst_wasapi2_result (hr)) {
+ GST_INFO_OBJECT (self, "Couldn't get period");
+ return hr;
+ }
+
+ GST_INFO_OBJECT (self, "Using IAudioClient3, default period %d frames, "
+ "fundamental period %d frames, minimum period %d frames, maximum period "
+ "%d frames", default_period, fundamental_period, min_period, max_period);
+
+ *period = min_period;
+
+ hr = audio_client->InitializeSharedAudioStream (stream_flags, min_period,
+ mix_format, nullptr);
+
+ if (!gst_wasapi2_result (hr))
+ GST_WARNING_OBJECT (self, "Failed to initialize IAudioClient3");
+
+ return hr;
+}
+
+static HRESULT
+gst_wasapi2_ring_buffer_initialize_audio_client (GstWasapi2RingBuffer * self,
+ IAudioClient * client_handle, WAVEFORMATEX * mix_format, guint * period)
+{
+ GstAudioRingBuffer *ringbuffer = GST_AUDIO_RING_BUFFER_CAST (self);
+ REFERENCE_TIME default_period, min_period;
+ DWORD stream_flags =
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK | AUDCLNT_STREAMFLAGS_NOPERSIST;
+ HRESULT hr;
+
+ hr = client_handle->GetDevicePeriod (&default_period, &min_period);
+ if (!gst_wasapi2_result (hr)) {
+ GST_WARNING_OBJECT (self, "Couldn't get device period info");
+ return hr;
+ }
+
+ GST_INFO_OBJECT (self, "wasapi2 default period: %" G_GINT64_FORMAT
+ ", min period: %" G_GINT64_FORMAT, default_period, min_period);
+
+ hr = client_handle->Initialize (AUDCLNT_SHAREMODE_SHARED, stream_flags,
+ /* hnsBufferDuration should be same as hnsPeriodicity
+ * when AUDCLNT_STREAMFLAGS_EVENTCALLBACK is used.
+ * And in case of shared mode, hnsPeriodicity should be zero, so
+ * this value should be zero as well */
+ 0,
+ /* This must always be 0 in shared mode */
+ 0, mix_format, nullptr);
+
+ if (!gst_wasapi2_result (hr)) {
+ GST_WARNING_OBJECT (self, "Couldn't initialize audioclient");
+ return hr;
+ }
+
+ *period = gst_util_uint64_scale_round (default_period * 100,
+ GST_AUDIO_INFO_RATE (&ringbuffer->spec.info), GST_SECOND);
+
+ return S_OK;
+}
+
+static gboolean
+gst_wasapi2_ring_buffer_acquire (GstAudioRingBuffer * buf,
+ GstAudioRingBufferSpec * spec)
+{
+ GstWasapi2RingBuffer *self = GST_WASAPI2_RING_BUFFER (buf);
+ IAudioClient *client_handle;
+ HRESULT hr;
+ WAVEFORMATEX *mix_format = nullptr;
+ ComPtr < ISimpleAudioVolume > audio_volume;
+ GstAudioChannelPosition *position = nullptr;
+ guint period = 0;
+
+ GST_DEBUG_OBJECT (buf, "Acquire");
+
+ if (!self->client) {
+ GST_ERROR_OBJECT (self, "No configured client object");
+ return FALSE;
+ }
+
+ if (!gst_wasapi2_client_ensure_activation (self->client)) {
+ GST_ERROR_OBJECT (self, "Failed to activate audio client");
+ return FALSE;
+ }
+
+ client_handle = gst_wasapi2_client_get_handle (self->client);
+ if (!client_handle) {
+ GST_ERROR_OBJECT (self, "IAudioClient handle is not available");
+ return FALSE;
+ }
+
+ /* TODO: convert given caps to mix format */
+ hr = client_handle->GetMixFormat (&mix_format);
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "Failed to get mix format");
+ return FALSE;
+ }
+
+ /* Only use audioclient3 when low-latency is requested because otherwise
+ * very slow machines and VMs with 1 CPU allocated will get glitches:
+ * https://bugzilla.gnome.org/show_bug.cgi?id=794497 */
+ hr = E_FAIL;
+ if (self->low_latency) {
+ hr = gst_wasapi2_ring_buffer_initialize_audio_client3 (self, client_handle,
+ mix_format, &period);
+ }
+
+ /* Try again if IAudioClinet3 API is unavailable.
+ * NOTE: IAudioClinet3:: methods might not be available for default device
+ * NOTE: The default device is a special device which is needed for supporting
+ * automatic stream routing
+ * https://docs.microsoft.com/en-us/windows/win32/coreaudio/automatic-stream-routing
+ */
+ if (FAILED (hr)) {
+ hr = gst_wasapi2_ring_buffer_initialize_audio_client (self, client_handle,
+ mix_format, &period);
+ }
+
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "Failed to initialize audio client");
+ return FALSE;
+ }
+
+ hr = client_handle->SetEventHandle (self->event_handle);
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "Failed to set event handle");
+ return FALSE;
+ }
+
+ gst_wasapi2_util_waveformatex_to_channel_mask (mix_format, &position);
+ if (position)
+ gst_audio_ring_buffer_set_channel_positions (buf, position);
+ g_free (position);
+
+ CoTaskMemFree (mix_format);
+
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "Failed to init audio client");
+ return FALSE;
+ }
+
+ hr = client_handle->GetBufferSize (&self->buffer_size);
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "Failed to query buffer size");
+ return FALSE;
+ }
+
+ g_assert (period > 0);
+
+ spec->segsize = period * GST_AUDIO_INFO_BPF (&buf->spec.info);
+ spec->segtotal = 2;
+
+ GST_INFO_OBJECT (self,
+ "Buffer size: %d frames, period: %d frames, segsize: %d bytes",
+ self->buffer_size, period, spec->segsize);
+
+ if (self->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER) {
+ ComPtr < IAudioRenderClient > render_client;
+
+ hr = client_handle->GetService (IID_PPV_ARGS (&render_client));
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "IAudioRenderClient is unavailable");
+ return FALSE;
+ }
+
+ self->render_client = render_client.Detach ();
+ } else {
+ ComPtr < IAudioCaptureClient > capture_client;
+
+ hr = client_handle->GetService (IID_PPV_ARGS (&capture_client));
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "IAudioCaptureClient is unavailable");
+ return FALSE;
+ }
+
+ self->capture_client = capture_client.Detach ();
+ }
+
+ hr = client_handle->GetService (IID_PPV_ARGS (&audio_volume));
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "ISimpleAudioVolume is unavailable");
+ goto error;
+ }
+
+ g_mutex_lock (&self->volume_lock);
+ self->volume_object = audio_volume.Detach ();
+
+ if (self->mute_changed) {
+ self->volume_object->SetMute (self->mute, nullptr);
+ self->mute_changed = FALSE;
+ } else {
+ self->volume_object->SetMute (FALSE, nullptr);
+ }
+
+ if (self->volume_changed) {
+ self->volume_object->SetMasterVolume (self->volume, nullptr);
+ self->volume_changed = FALSE;
+ }
+ g_mutex_unlock (&self->volume_lock);
+
+ buf->size = spec->segtotal * spec->segsize;
+ buf->memory = (guint8 *) g_malloc (buf->size);
+ gst_audio_format_info_fill_silence (buf->spec.info.finfo,
+ buf->memory, buf->size);
+
+ return TRUE;
+
+error:
+ GST_WASAPI2_CLEAR_COM (self->render_client);
+ GST_WASAPI2_CLEAR_COM (self->capture_client);
+ GST_WASAPI2_CLEAR_COM (self->volume_object);
+
+ return FALSE;
+}
+
+static gboolean
+gst_wasapi2_ring_buffer_release (GstAudioRingBuffer * buf)
+{
+ GST_DEBUG_OBJECT (buf, "Release");
+
+ g_clear_pointer (&buf->memory, g_free);
+
+ return TRUE;
+}
+
+static gboolean
+gst_wasapi2_ring_buffer_start (GstAudioRingBuffer * buf)
+{
+ GstWasapi2RingBuffer *self = GST_WASAPI2_RING_BUFFER (buf);
+ IAudioClient *client_handle;
+ HRESULT hr;
+
+ GST_DEBUG_OBJECT (self, "Start");
+
+ client_handle = gst_wasapi2_client_get_handle (self->client);
+ self->is_first = TRUE;
+ self->running = TRUE;
+ self->segoffset = 0;
+ self->write_frame_offset = 0;
+
+ /* render client might read data from buffer immediately once it's prepared.
+ * Pre-fill with silence in order to start-up glitch */
+ if (self->device_class == GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER) {
+ hr = gst_wasapi2_ring_buffer_write (self, TRUE);
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "Failed to pre-fill buffer with silence");
+ return FALSE;
+ }
+ }
+
+ hr = client_handle->Start ();
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "Failed to start client");
+ self->running = FALSE;
+ return FALSE;
+ }
+
+ hr = MFPutWaitingWorkItem (self->event_handle, 0, self->callback_result,
+ &self->callback_key);
+ if (!gst_wasapi2_result (hr)) {
+ GST_ERROR_OBJECT (self, "Failed to put waiting item");
+ client_handle->Stop ();
+ self->running = FALSE;
+
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+static gboolean
+gst_wasapi2_ring_buffer_stop (GstAudioRingBuffer * buf)
+{
+ GstWasapi2RingBuffer *self = GST_WASAPI2_RING_BUFFER (buf);
+ IAudioClient *client_handle;
+ HRESULT hr;
+
+ GST_DEBUG_OBJECT (buf, "Stop");
+
+ if (!self->client) {
+ GST_DEBUG_OBJECT (self, "No configured client");
+ return TRUE;
+ }
+
+ if (!self->running) {
+ GST_DEBUG_OBJECT (self, "We are not running");
+ return TRUE;
+ }
+
+ client_handle = gst_wasapi2_client_get_handle (self->client);
+
+ self->running = FALSE;
+ MFCancelWorkItem (self->callback_key);
+
+ hr = client_handle->Stop ();
+ gst_wasapi2_result (hr);
+
+ /* Call reset for later reuse case */
+ hr = client_handle->Reset ();
+ self->expected_position = 0;
+
+ return TRUE;
+}
+
+static guint
+gst_wasapi2_ring_buffer_delay (GstAudioRingBuffer * buf)
+{
+ /* NOTE: WASAPI supports GetCurrentPadding() method for querying
+ * currently unread buffer size, but it doesn't seem to be quite useful
+ * here because:
+ *
+ * In case of capture client, GetCurrentPadding() will return the number of
+ * unread frames which will be identical to pNumFramesToRead value of
+ * IAudioCaptureClient::GetBuffer()'s return. Since we are running on
+ * event-driven mode and whenever available, WASAPI will notify signal
+ * so it's likely zero at this moment. And there is a chance to
+ * return incorrect value here because our IO callback happens from
+ * other thread.
+ *
+ * And render client's padding size will return the total size of buffer
+ * which is likely larger than twice of our period. Which doesn't represent
+ * the amount queued frame size in device correctly
+ */
+ return 0;
+}
+
+GstAudioRingBuffer *
+gst_wasapi2_ring_buffer_new (GstWasapi2ClientDeviceClass device_class,
+ gboolean low_latency, const gchar * device_id, gpointer dispatcher,
+ const gchar * name)
+{
+ GstWasapi2RingBuffer *self;
+
+ self = (GstWasapi2RingBuffer *)
+ g_object_new (GST_TYPE_WASAPI2_RING_BUFFER, "name", name, nullptr);
+
+ if (!self->callback_object) {
+ gst_object_unref (self);
+ return nullptr;
+ }
+
+ self->device_class = device_class;
+ self->low_latency = low_latency;
+ self->device_id = g_strdup (device_id);
+ self->dispatcher = dispatcher;
+
+ return GST_AUDIO_RING_BUFFER_CAST (self);
+}
+
+GstCaps *
+gst_wasapi2_ring_buffer_get_caps (GstWasapi2RingBuffer * buf)
+{
+ g_return_val_if_fail (GST_IS_WASAPI2_RING_BUFFER (buf), nullptr);
+
+ if (!buf->client)
+ return nullptr;
+
+ if (!gst_wasapi2_client_ensure_activation (buf->client)) {
+ GST_ERROR_OBJECT (buf, "Failed to activate audio client");
+ return nullptr;
+ }
+
+ return gst_wasapi2_client_get_caps (buf->client);
+}
+
+HRESULT
+gst_wasapi2_ring_buffer_set_mute (GstWasapi2RingBuffer * buf, gboolean mute)
+{
+ HRESULT hr = S_OK;
+ g_return_val_if_fail (GST_IS_WASAPI2_RING_BUFFER (buf), E_INVALIDARG);
+
+ g_mutex_lock (&buf->volume_lock);
+ buf->mute = mute;
+ if (buf->volume_object)
+ hr = buf->volume_object->SetMute (mute, nullptr);
+ else
+ buf->volume_changed = TRUE;
+ g_mutex_unlock (&buf->volume_lock);
+
+ return S_OK;
+}
+
+HRESULT
+gst_wasapi2_ring_buffer_get_mute (GstWasapi2RingBuffer * buf, gboolean * mute)
+{
+ BOOL mute_val;
+ HRESULT hr = S_OK;
+
+ g_return_val_if_fail (GST_IS_WASAPI2_RING_BUFFER (buf), E_INVALIDARG);
+ g_return_val_if_fail (mute != nullptr, E_INVALIDARG);
+
+ mute_val = buf->mute;
+
+ g_mutex_lock (&buf->volume_lock);
+ if (buf->volume_object)
+ hr = buf->volume_object->GetMute (&mute_val);
+ g_mutex_unlock (&buf->volume_lock);
+
+ *mute = mute_val ? TRUE : FALSE;
+
+ return hr;
+}
+
+HRESULT
+gst_wasapi2_ring_buffer_set_volume (GstWasapi2RingBuffer * buf, gfloat volume)
+{
+ HRESULT hr = S_OK;
+
+ g_return_val_if_fail (GST_IS_WASAPI2_RING_BUFFER (buf), E_INVALIDARG);
+ g_return_val_if_fail (volume >= 0 && volume <= 1.0, E_INVALIDARG);
+
+ g_mutex_lock (&buf->volume_lock);
+ buf->volume = volume;
+ if (buf->volume_object)
+ hr = buf->volume_object->SetMasterVolume (volume, nullptr);
+ else
+ buf->mute_changed = TRUE;
+ g_mutex_unlock (&buf->volume_lock);
+
+ return hr;
+}
+
+HRESULT
+gst_wasapi2_ring_buffer_get_volume (GstWasapi2RingBuffer * buf, gfloat * volume)
+{
+ gfloat volume_val;
+ HRESULT hr = S_OK;
+
+ g_return_val_if_fail (GST_IS_WASAPI2_RING_BUFFER (buf), E_INVALIDARG);
+ g_return_val_if_fail (volume != nullptr, E_INVALIDARG);
+
+ g_mutex_lock (&buf->volume_lock);
+ volume_val = buf->volume;
+ if (buf->volume_object)
+ hr = buf->volume_object->GetMasterVolume (&volume_val);
+ g_mutex_unlock (&buf->volume_lock);
+
+ *volume = volume_val;
+
+ return hr;
+}
diff --git a/sys/wasapi2/gstwasapi2ringbuffer.h b/sys/wasapi2/gstwasapi2ringbuffer.h
new file mode 100644
index 000000000..5bbb6e976
--- /dev/null
+++ b/sys/wasapi2/gstwasapi2ringbuffer.h
@@ -0,0 +1,55 @@
+/* GStreamer
+ * Copyright (C) 2021 Seungha Yang <seungha@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.
+ */
+
+#ifndef __GST_WASAPI2_RING_BUFFER_H__
+#define __GST_WASAPI2_RING_BUFFER_H__
+
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+#include "gstwasapi2client.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_WASAPI2_RING_BUFFER (gst_wasapi2_ring_buffer_get_type())
+G_DECLARE_FINAL_TYPE (GstWasapi2RingBuffer, gst_wasapi2_ring_buffer,
+ GST, WASAPI2_RING_BUFFER, GstAudioRingBuffer);
+
+GstAudioRingBuffer * gst_wasapi2_ring_buffer_new (GstWasapi2ClientDeviceClass device_class,
+ gboolean low_latency,
+ const gchar *device_id,
+ gpointer dispatcher,
+ const gchar * name);
+
+GstCaps * gst_wasapi2_ring_buffer_get_caps (GstWasapi2RingBuffer * buf);
+
+HRESULT gst_wasapi2_ring_buffer_set_mute (GstWasapi2RingBuffer * buf,
+ gboolean mute);
+
+HRESULT gst_wasapi2_ring_buffer_get_mute (GstWasapi2RingBuffer * buf,
+ gboolean * mute);
+
+HRESULT gst_wasapi2_ring_buffer_set_volume (GstWasapi2RingBuffer * buf,
+ gfloat volume);
+
+HRESULT gst_wasapi2_ring_buffer_get_volume (GstWasapi2RingBuffer * buf,
+ gfloat * volume);
+
+G_END_DECLS
+
+#endif /* __GST_WASAPI2_RING_BUFFER_H__ */
diff --git a/sys/wasapi2/gstwasapi2sink.c b/sys/wasapi2/gstwasapi2sink.c
index cb3fd407f..3bb82529e 100644
--- a/sys/wasapi2/gstwasapi2sink.c
+++ b/sys/wasapi2/gstwasapi2sink.c
@@ -31,11 +31,11 @@
*
* ## Example pipelines
* |[
- * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapi2sink
+ * gst-launch-1.0 -v audiotestsink samplesperbuffer=160 ! wasapi2sink
* ]| Generate 20 ms buffers and render to the default audio device.
*
* |[
- * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapi2sink low-latency=true
+ * gst-launch-1.0 -v audiotestsink samplesperbuffer=160 ! wasapi2sink low-latency=true
* ]| Same as above, but with the minimum possible latency
*
*/
@@ -45,7 +45,7 @@
#include "gstwasapi2sink.h"
#include "gstwasapi2util.h"
-#include "gstwasapi2client.h"
+#include "gstwasapi2ringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_wasapi2_sink_debug);
#define GST_CAT_DEFAULT gst_wasapi2_sink_debug
@@ -59,9 +59,6 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
#define DEFAULT_MUTE FALSE
#define DEFAULT_VOLUME 1.0
-#define GST_WASAPI2_SINK_LOCK(s) g_mutex_lock(&(s)->lock)
-#define GST_WASAPI2_SINK_UNLOCK(s) g_mutex_unlock(&(s)->lock)
-
enum
{
PROP_0,
@@ -74,11 +71,7 @@ enum
struct _GstWasapi2Sink
{
- GstAudioSink parent;
-
- GstWasapi2Client *client;
- GstCaps *cached_caps;
- gboolean started;
+ GstAudioBaseSink parent;
/* properties */
gchar *device_id;
@@ -89,30 +82,21 @@ struct _GstWasapi2Sink
gboolean mute_changed;
gboolean volume_changed;
-
- /* to protect audioclient from set/get property */
- GMutex lock;
};
-static void gst_wasapi2_sink_dispose (GObject * object);
static void gst_wasapi2_sink_finalize (GObject * object);
static void gst_wasapi2_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wasapi2_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
+static GstStateChangeReturn gst_wasapi2_sink_change_state (GstElement *
+ element, GstStateChange transition);
+
static GstCaps *gst_wasapi2_sink_get_caps (GstBaseSink * bsink,
GstCaps * filter);
-
-static gboolean gst_wasapi2_sink_prepare (GstAudioSink * asink,
- GstAudioRingBufferSpec * spec);
-static gboolean gst_wasapi2_sink_unprepare (GstAudioSink * asink);
-static gboolean gst_wasapi2_sink_open (GstAudioSink * asink);
-static gboolean gst_wasapi2_sink_close (GstAudioSink * asink);
-static gint gst_wasapi2_sink_write (GstAudioSink * asink,
- gpointer data, guint length);
-static guint gst_wasapi2_sink_delay (GstAudioSink * asink);
-static void gst_wasapi2_sink_reset (GstAudioSink * asink);
+static GstAudioRingBuffer *gst_wasapi2_sink_create_ringbuffer (GstAudioBaseSink
+ * sink);
static void gst_wasapi2_sink_set_mute (GstWasapi2Sink * self, gboolean mute);
static gboolean gst_wasapi2_sink_get_mute (GstWasapi2Sink * self);
@@ -120,7 +104,8 @@ static void gst_wasapi2_sink_set_volume (GstWasapi2Sink * self, gdouble volume);
static gdouble gst_wasapi2_sink_get_volume (GstWasapi2Sink * self);
#define gst_wasapi2_sink_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstWasapi2Sink, gst_wasapi2_sink, GST_TYPE_AUDIO_SINK,
+G_DEFINE_TYPE_WITH_CODE (GstWasapi2Sink, gst_wasapi2_sink,
+ GST_TYPE_AUDIO_BASE_SINK,
G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL));
static void
@@ -129,9 +114,9 @@ gst_wasapi2_sink_class_init (GstWasapi2SinkClass * klass)
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass);
- GstAudioSinkClass *audiosink_class = GST_AUDIO_SINK_CLASS (klass);
+ GstAudioBaseSinkClass *audiobasesink_class =
+ GST_AUDIO_BASE_SINK_CLASS (klass);
- gobject_class->dispose = gst_wasapi2_sink_dispose;
gobject_class->finalize = gst_wasapi2_sink_finalize;
gobject_class->set_property = gst_wasapi2_sink_set_property;
gobject_class->get_property = gst_wasapi2_sink_get_property;
@@ -184,15 +169,13 @@ gst_wasapi2_sink_class_init (GstWasapi2SinkClass * klass)
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>, "
"Seungha Yang <seungha@centricular.com>");
+ element_class->change_state =
+ GST_DEBUG_FUNCPTR (gst_wasapi2_sink_change_state);
+
basesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_get_caps);
- audiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_prepare);
- audiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_unprepare);
- audiosink_class->open = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_open);
- audiosink_class->close = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_close);
- audiosink_class->write = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_write);
- audiosink_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_delay);
- audiosink_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi2_sink_reset);
+ audiobasesink_class->create_ringbuffer =
+ GST_DEBUG_FUNCPTR (gst_wasapi2_sink_create_ringbuffer);
GST_DEBUG_CATEGORY_INIT (gst_wasapi2_sink_debug, "wasapi2sink",
0, "Windows audio session API sink");
@@ -204,21 +187,6 @@ gst_wasapi2_sink_init (GstWasapi2Sink * self)
self->low_latency = DEFAULT_LOW_LATENCY;
self->mute = DEFAULT_MUTE;
self->volume = DEFAULT_VOLUME;
-
- g_mutex_init (&self->lock);
-}
-
-static void
-gst_wasapi2_sink_dispose (GObject * object)
-{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (object);
-
- GST_WASAPI2_SINK_LOCK (self);
- gst_clear_object (&self->client);
- gst_clear_caps (&self->cached_caps);
- GST_WASAPI2_SINK_UNLOCK (self);
-
- G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
@@ -227,7 +195,6 @@ gst_wasapi2_sink_finalize (GObject * object)
GstWasapi2Sink *self = GST_WASAPI2_SINK (object);
g_free (self->device_id);
- g_mutex_clear (&self->lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
@@ -286,29 +253,58 @@ gst_wasapi2_sink_get_property (GObject * object, guint prop_id,
}
}
+static GstStateChangeReturn
+gst_wasapi2_sink_change_state (GstElement * element, GstStateChange transition)
+{
+ GstWasapi2Sink *self = GST_WASAPI2_SINK (element);
+ GstAudioBaseSink *asink = GST_AUDIO_BASE_SINK_CAST (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ /* If we have pending volume/mute values to set, do here */
+ GST_OBJECT_LOCK (self);
+ if (asink->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (asink->ringbuffer);
+
+ if (self->volume_changed) {
+ gst_wasapi2_ring_buffer_set_volume (ringbuffer, self->volume);
+ self->volume_changed = FALSE;
+ }
+
+ if (self->mute_changed) {
+ gst_wasapi2_ring_buffer_set_mute (ringbuffer, self->mute);
+ self->mute_changed = FALSE;
+ }
+ }
+ GST_OBJECT_UNLOCK (self);
+ break;
+ default:
+ break;
+ }
+
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+}
+
static GstCaps *
gst_wasapi2_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (bsink);
+ GstAudioBaseSink *asink = GST_AUDIO_BASE_SINK_CAST (bsink);
GstCaps *caps = NULL;
- /* In case of UWP, device activation might not be finished yet */
- if (self->client && !gst_wasapi2_client_ensure_activation (self->client)) {
- GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
- ("Failed to activate device"));
- return NULL;
- }
+ GST_OBJECT_LOCK (bsink);
+ if (asink->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (asink->ringbuffer);
- if (self->client)
- caps = gst_wasapi2_client_get_caps (self->client);
+ gst_object_ref (ringbuffer);
+ GST_OBJECT_UNLOCK (bsink);
- /* store one caps here so that we can return device caps even if
- * audioclient was closed due to unprepare() */
- if (!self->cached_caps && caps)
- self->cached_caps = gst_caps_ref (caps);
-
- if (!caps && self->cached_caps)
- caps = gst_caps_ref (self->cached_caps);
+ /* Get caps might be able to block if device is not activated yet */
+ caps = gst_wasapi2_ring_buffer_get_caps (ringbuffer);
+ } else {
+ GST_OBJECT_UNLOCK (bsink);
+ }
if (!caps)
caps = gst_pad_get_pad_template_caps (bsink->sinkpad);
@@ -320,238 +316,81 @@ gst_wasapi2_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
caps = filtered;
}
- GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps);
+ GST_DEBUG_OBJECT (bsink, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
-static gboolean
-gst_wasapi2_sink_open_unlocked (GstAudioSink * asink)
-{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (asink);
-
- gst_clear_object (&self->client);
-
- self->client =
- gst_wasapi2_client_new (GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER,
- self->low_latency, -1, self->device_id, self->dispatcher);
-
- if (!self->client)
- return FALSE;
-
- return TRUE;
-}
-
-static gboolean
-gst_wasapi2_sink_open (GstAudioSink * asink)
-{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (asink);
- gboolean ret;
-
- GST_DEBUG_OBJECT (self, "Opening device");
-
- GST_WASAPI2_SINK_LOCK (self);
- ret = gst_wasapi2_sink_open_unlocked (asink);
- GST_WASAPI2_SINK_UNLOCK (self);
-
- if (!ret) {
- GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
- ("Failed to open device"));
- return FALSE;
- }
-
- return TRUE;
-}
-
-static gboolean
-gst_wasapi2_sink_close (GstAudioSink * asink)
-{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (asink);
-
- GST_WASAPI2_SINK_LOCK (self);
-
- gst_clear_object (&self->client);
- gst_clear_caps (&self->cached_caps);
- self->started = FALSE;
-
- GST_WASAPI2_SINK_UNLOCK (self);
-
- return TRUE;
-}
-
-static gboolean
-gst_wasapi2_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
-{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (asink);
- GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK (asink);
- gboolean ret = FALSE;
- HRESULT hr;
-
- GST_WASAPI2_SINK_LOCK (self);
- if (!self->client && !gst_wasapi2_sink_open_unlocked (asink)) {
- GST_ERROR_OBJECT (self, "No audio client was configured");
- goto done;
- }
-
- if (!gst_wasapi2_client_ensure_activation (self->client)) {
- GST_ERROR_OBJECT (self, "Couldn't activate audio device");
- goto done;
- }
-
- hr = gst_wasapi2_client_open (self->client, spec, bsink->ringbuffer);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (self, "Couldn't open audio client");
- goto done;
- }
-
- /* Set mute and volume here again, maybe when "mute" property was set, audioclient
- * might not be configured at that moment */
- if (self->mute_changed) {
- gst_wasapi2_client_set_mute (self->client, self->mute);
- self->mute_changed = FALSE;
- }
-
- if (self->volume_changed) {
- gst_wasapi2_client_set_volume (self->client, self->volume);
- self->volume_changed = FALSE;
- }
-
- /* Will start IAudioClient on the first write request */
- self->started = FALSE;
- ret = TRUE;
-
-done:
- GST_WASAPI2_SINK_UNLOCK (self);
-
- return ret;
-}
-
-static gboolean
-gst_wasapi2_sink_unprepare (GstAudioSink * asink)
-{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (asink);
-
- self->started = FALSE;
-
- /* Will reopen device later prepare() */
- GST_WASAPI2_SINK_LOCK (self);
- if (self->client) {
- gst_wasapi2_client_stop (self->client);
- gst_clear_object (&self->client);
- }
- GST_WASAPI2_SINK_UNLOCK (self);
-
- return TRUE;
-}
-
-static gint
-gst_wasapi2_sink_write (GstAudioSink * asink, gpointer data, guint length)
+static GstAudioRingBuffer *
+gst_wasapi2_sink_create_ringbuffer (GstAudioBaseSink * sink)
{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (asink);
- HRESULT hr;
- guint write_len = 0;
+ GstWasapi2Sink *self = GST_WASAPI2_SINK (sink);
+ GstAudioRingBuffer *ringbuffer;
+ gchar *name;
- if (!self->client) {
- GST_ERROR_OBJECT (self, "No audio client was configured");
- return -1;
- }
+ name = g_strdup_printf ("%s-ringbuffer", GST_OBJECT_NAME (sink));
- if (!self->started) {
- HRESULT hr = gst_wasapi2_client_start (self->client);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (self, "Failed to re-start client");
- return -1;
- }
+ ringbuffer =
+ gst_wasapi2_ring_buffer_new (GST_WASAPI2_CLIENT_DEVICE_CLASS_RENDER,
+ self->low_latency, self->device_id, self->dispatcher, name);
- self->started = TRUE;
- }
+ g_free (name);
- hr = gst_wasapi2_client_write (self->client, data, length, &write_len);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (self, "Failed to write");
- return -1;
- }
-
- return (gint) write_len;
-}
-
-static guint
-gst_wasapi2_sink_delay (GstAudioSink * asink)
-{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (asink);
- guint32 delay;
- HRESULT hr;
-
- if (!self->client)
- return 0;
-
- hr = gst_wasapi2_client_delay (self->client, &delay);
- if (!gst_wasapi2_result (hr)) {
- GST_WARNING_OBJECT (self, "Failed to get delay");
- return 0;
- }
-
- return delay;
-}
-
-static void
-gst_wasapi2_sink_reset (GstAudioSink * asink)
-{
- GstWasapi2Sink *self = GST_WASAPI2_SINK (asink);
-
- GST_INFO_OBJECT (self, "reset called");
-
- self->started = FALSE;
-
- if (!self->client)
- return;
-
- gst_wasapi2_client_stop (self->client);
+ return ringbuffer;
}
static void
gst_wasapi2_sink_set_mute (GstWasapi2Sink * self, gboolean mute)
{
- GST_WASAPI2_SINK_LOCK (self);
+ GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK_CAST (self);
+ HRESULT hr;
+
+ GST_OBJECT_LOCK (self);
self->mute = mute;
self->mute_changed = TRUE;
- if (self->client) {
- HRESULT hr = gst_wasapi2_client_set_mute (self->client, mute);
+ if (bsink->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (bsink->ringbuffer);
+
+ hr = gst_wasapi2_ring_buffer_set_mute (ringbuffer, mute);
+
if (FAILED (hr)) {
GST_INFO_OBJECT (self, "Couldn't set mute");
} else {
self->mute_changed = FALSE;
}
- } else {
- GST_DEBUG_OBJECT (self, "audio client is not configured yet");
}
- GST_WASAPI2_SINK_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
}
static gboolean
gst_wasapi2_sink_get_mute (GstWasapi2Sink * self)
{
+ GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK_CAST (self);
gboolean mute;
+ HRESULT hr;
- GST_WASAPI2_SINK_LOCK (self);
+ GST_OBJECT_LOCK (self);
mute = self->mute;
- if (self->client) {
- HRESULT hr = gst_wasapi2_client_get_mute (self->client, &mute);
+ if (bsink->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (bsink->ringbuffer);
+
+ hr = gst_wasapi2_ring_buffer_get_mute (ringbuffer, &mute);
+
if (FAILED (hr)) {
- GST_INFO_OBJECT (self, "Couldn't get mute state");
+ GST_INFO_OBJECT (self, "Couldn't get mute");
} else {
self->mute = mute;
}
- } else {
- GST_DEBUG_OBJECT (self, "audio client is not configured yet");
}
- GST_WASAPI2_SINK_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
return mute;
}
@@ -559,7 +398,10 @@ gst_wasapi2_sink_get_mute (GstWasapi2Sink * self)
static void
gst_wasapi2_sink_set_volume (GstWasapi2Sink * self, gdouble volume)
{
- GST_WASAPI2_SINK_LOCK (self);
+ GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK_CAST (self);
+ HRESULT hr;
+
+ GST_OBJECT_LOCK (self);
self->volume = volume;
/* clip volume value */
@@ -567,42 +409,47 @@ gst_wasapi2_sink_set_volume (GstWasapi2Sink * self, gdouble volume)
self->volume = MIN (1.0, self->volume);
self->volume_changed = TRUE;
- if (self->client) {
- HRESULT hr =
- gst_wasapi2_client_set_volume (self->client, (gfloat) self->volume);
+ if (bsink->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (bsink->ringbuffer);
+
+ hr = gst_wasapi2_ring_buffer_set_volume (ringbuffer, (gfloat) self->volume);
+
if (FAILED (hr)) {
GST_INFO_OBJECT (self, "Couldn't set volume");
} else {
self->volume_changed = FALSE;
}
- } else {
- GST_DEBUG_OBJECT (self, "audio client is not configured yet");
}
- GST_WASAPI2_SINK_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
}
static gdouble
gst_wasapi2_sink_get_volume (GstWasapi2Sink * self)
{
+ GstAudioBaseSink *bsink = GST_AUDIO_BASE_SINK_CAST (self);
gfloat volume;
+ HRESULT hr;
- GST_WASAPI2_SINK_LOCK (self);
+ GST_OBJECT_LOCK (self);
volume = (gfloat) self->volume;
- if (self->client) {
- HRESULT hr = gst_wasapi2_client_get_volume (self->client, &volume);
+ if (bsink->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (bsink->ringbuffer);
+
+ hr = gst_wasapi2_ring_buffer_get_volume (ringbuffer, &volume);
+
if (FAILED (hr)) {
- GST_INFO_OBJECT (self, "Couldn't get volume");
+ GST_INFO_OBJECT (self, "Couldn't set volume");
} else {
self->volume = volume;
}
- } else {
- GST_DEBUG_OBJECT (self, "audio client is not configured yet");
}
- GST_WASAPI2_SINK_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
volume = MAX (0.0, volume);
volume = MIN (1.0, volume);
diff --git a/sys/wasapi2/gstwasapi2sink.h b/sys/wasapi2/gstwasapi2sink.h
index 433dcb656..5e618379e 100644
--- a/sys/wasapi2/gstwasapi2sink.h
+++ b/sys/wasapi2/gstwasapi2sink.h
@@ -27,7 +27,7 @@ G_BEGIN_DECLS
#define GST_TYPE_WASAPI2_SINK (gst_wasapi2_sink_get_type ())
G_DECLARE_FINAL_TYPE (GstWasapi2Sink,
- gst_wasapi2_sink, GST, WASAPI2_SINK, GstAudioSink);
+ gst_wasapi2_sink, GST, WASAPI2_SINK, GstAudioBaseSink);
G_END_DECLS
diff --git a/sys/wasapi2/gstwasapi2src.c b/sys/wasapi2/gstwasapi2src.c
index fa5616936..446803aac 100644
--- a/sys/wasapi2/gstwasapi2src.c
+++ b/sys/wasapi2/gstwasapi2src.c
@@ -43,7 +43,7 @@
#include "gstwasapi2src.h"
#include "gstwasapi2util.h"
-#include "gstwasapi2client.h"
+#include "gstwasapi2ringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_wasapi2_src_debug);
#define GST_CAT_DEFAULT gst_wasapi2_src_debug
@@ -57,9 +57,6 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
#define DEFAULT_MUTE FALSE
#define DEFAULT_VOLUME 1.0
-#define GST_WASAPI2_SRC_LOCK(s) g_mutex_lock(&(s)->lock)
-#define GST_WASAPI2_SRC_UNLOCK(s) g_mutex_unlock(&(s)->lock)
-
enum
{
PROP_0,
@@ -72,11 +69,7 @@ enum
struct _GstWasapi2Src
{
- GstAudioSrc parent;
-
- GstWasapi2Client *client;
- GstCaps *cached_caps;
- gboolean started;
+ GstAudioBaseSrc parent;
/* properties */
gchar *device_id;
@@ -87,29 +80,20 @@ struct _GstWasapi2Src
gboolean mute_changed;
gboolean volume_changed;
-
- /* to protect audioclient from set/get property */
- GMutex lock;
};
-static void gst_wasapi2_src_dispose (GObject * object);
static void gst_wasapi2_src_finalize (GObject * object);
static void gst_wasapi2_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wasapi2_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-static GstCaps *gst_wasapi2_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
+static GstStateChangeReturn gst_wasapi2_src_change_state (GstElement *
+ element, GstStateChange transition);
-static gboolean gst_wasapi2_src_open (GstAudioSrc * asrc);
-static gboolean gst_wasapi2_src_close (GstAudioSrc * asrc);
-static gboolean gst_wasapi2_src_prepare (GstAudioSrc * asrc,
- GstAudioRingBufferSpec * spec);
-static gboolean gst_wasapi2_src_unprepare (GstAudioSrc * asrc);
-static guint gst_wasapi2_src_read (GstAudioSrc * asrc, gpointer data,
- guint length, GstClockTime * timestamp);
-static guint gst_wasapi2_src_delay (GstAudioSrc * asrc);
-static void gst_wasapi2_src_reset (GstAudioSrc * asrc);
+static GstCaps *gst_wasapi2_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
+static GstAudioRingBuffer *gst_wasapi2_src_create_ringbuffer (GstAudioBaseSrc *
+ src);
static void gst_wasapi2_src_set_mute (GstWasapi2Src * self, gboolean mute);
static gboolean gst_wasapi2_src_get_mute (GstWasapi2Src * self);
@@ -117,7 +101,8 @@ static void gst_wasapi2_src_set_volume (GstWasapi2Src * self, gdouble volume);
static gdouble gst_wasapi2_src_get_volume (GstWasapi2Src * self);
#define gst_wasapi2_src_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstWasapi2Src, gst_wasapi2_src, GST_TYPE_AUDIO_SRC,
+G_DEFINE_TYPE_WITH_CODE (GstWasapi2Src, gst_wasapi2_src,
+ GST_TYPE_AUDIO_BASE_SRC,
G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL));
static void
@@ -126,9 +111,8 @@ gst_wasapi2_src_class_init (GstWasapi2SrcClass * klass)
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *basesrc_class = GST_BASE_SRC_CLASS (klass);
- GstAudioSrcClass *audiosrc_class = GST_AUDIO_SRC_CLASS (klass);
+ GstAudioBaseSrcClass *audiobasesrc_class = GST_AUDIO_BASE_SRC_CLASS (klass);
- gobject_class->dispose = gst_wasapi2_src_dispose;
gobject_class->finalize = gst_wasapi2_src_finalize;
gobject_class->set_property = gst_wasapi2_src_set_property;
gobject_class->get_property = gst_wasapi2_src_get_property;
@@ -181,15 +165,13 @@ gst_wasapi2_src_class_init (GstWasapi2SrcClass * klass)
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>, "
"Seungha Yang <seungha@centricular.com>");
+ element_class->change_state =
+ GST_DEBUG_FUNCPTR (gst_wasapi2_src_change_state);
+
basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi2_src_get_caps);
- audiosrc_class->open = GST_DEBUG_FUNCPTR (gst_wasapi2_src_open);
- audiosrc_class->close = GST_DEBUG_FUNCPTR (gst_wasapi2_src_close);
- audiosrc_class->read = GST_DEBUG_FUNCPTR (gst_wasapi2_src_read);
- audiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi2_src_prepare);
- audiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi2_src_unprepare);
- audiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi2_src_delay);
- audiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi2_src_reset);
+ audiobasesrc_class->create_ringbuffer =
+ GST_DEBUG_FUNCPTR (gst_wasapi2_src_create_ringbuffer);
GST_DEBUG_CATEGORY_INIT (gst_wasapi2_src_debug, "wasapi2src",
0, "Windows audio session API source");
@@ -201,21 +183,6 @@ gst_wasapi2_src_init (GstWasapi2Src * self)
self->mute = DEFAULT_MUTE;
self->volume = DEFAULT_VOLUME;
self->low_latency = DEFAULT_LOW_LATENCY;
-
- g_mutex_init (&self->lock);
-}
-
-static void
-gst_wasapi2_src_dispose (GObject * object)
-{
- GstWasapi2Src *self = GST_WASAPI2_SRC (object);
-
- GST_WASAPI2_SRC_LOCK (self);
- gst_clear_object (&self->client);
- gst_clear_caps (&self->cached_caps);
- GST_WASAPI2_SRC_UNLOCK (self);
-
- G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
@@ -224,7 +191,6 @@ gst_wasapi2_src_finalize (GObject * object)
GstWasapi2Src *self = GST_WASAPI2_SRC (object);
g_free (self->device_id);
- g_mutex_clear (&self->lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
@@ -283,29 +249,59 @@ gst_wasapi2_src_get_property (GObject * object, guint prop_id,
}
}
+static GstStateChangeReturn
+gst_wasapi2_src_change_state (GstElement * element, GstStateChange transition)
+{
+ GstWasapi2Src *self = GST_WASAPI2_SRC (element);
+ GstAudioBaseSrc *asrc = GST_AUDIO_BASE_SRC_CAST (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ /* If we have pending volume/mute values to set, do here */
+ GST_OBJECT_LOCK (self);
+ if (asrc->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (asrc->ringbuffer);
+
+ if (self->volume_changed) {
+ gst_wasapi2_ring_buffer_set_volume (ringbuffer, self->volume);
+ self->volume_changed = FALSE;
+ }
+
+ if (self->mute_changed) {
+ gst_wasapi2_ring_buffer_set_mute (ringbuffer, self->mute);
+ self->mute_changed = FALSE;
+ }
+ }
+ GST_OBJECT_UNLOCK (self);
+ break;
+ default:
+ break;
+ }
+
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+}
+
static GstCaps *
gst_wasapi2_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
{
- GstWasapi2Src *self = GST_WASAPI2_SRC (bsrc);
+ GstAudioBaseSrc *asrc = GST_AUDIO_BASE_SRC_CAST (bsrc);
GstCaps *caps = NULL;
- /* In case of UWP, device activation might not be finished yet */
- if (self->client && !gst_wasapi2_client_ensure_activation (self->client)) {
- GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL),
- ("Failed to activate device"));
- return NULL;
- }
+ GST_OBJECT_LOCK (bsrc);
+ if (asrc->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (asrc->ringbuffer);
- if (self->client)
- caps = gst_wasapi2_client_get_caps (self->client);
+ gst_object_ref (ringbuffer);
+ GST_OBJECT_UNLOCK (bsrc);
- /* store one caps here so that we can return device caps even if
- * audioclient was closed due to unprepare() */
- if (!self->cached_caps && caps)
- self->cached_caps = gst_caps_ref (caps);
-
- if (!caps && self->cached_caps)
- caps = gst_caps_ref (self->cached_caps);
+ /* Get caps might be able to block if device is not activated yet */
+ caps = gst_wasapi2_ring_buffer_get_caps (ringbuffer);
+ gst_object_unref (ringbuffer);
+ } else {
+ GST_OBJECT_UNLOCK (bsrc);
+ }
if (!caps)
caps = gst_pad_get_pad_template_caps (bsrc->srcpad);
@@ -317,239 +313,79 @@ gst_wasapi2_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
caps = filtered;
}
- GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps);
+ GST_DEBUG_OBJECT (bsrc, "returning caps %" GST_PTR_FORMAT, caps);
return caps;
}
-static gboolean
-gst_wasapi2_src_open_unlocked (GstAudioSrc * asrc)
-{
- GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
-
- gst_clear_object (&self->client);
-
- self->client =
- gst_wasapi2_client_new (GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE,
- self->low_latency, -1, self->device_id, self->dispatcher);
-
- if (!self->client)
- return FALSE;
-
- return TRUE;
-}
-
-static gboolean
-gst_wasapi2_src_open (GstAudioSrc * asrc)
-{
- GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
- gboolean ret;
-
- GST_DEBUG_OBJECT (self, "Opening device");
-
- GST_WASAPI2_SRC_LOCK (self);
- ret = gst_wasapi2_src_open_unlocked (asrc);
- GST_WASAPI2_SRC_UNLOCK (self);
-
- if (!ret) {
- GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
- ("Failed to open device"));
- return FALSE;
- }
-
- return TRUE;
-}
-
-static gboolean
-gst_wasapi2_src_close (GstAudioSrc * asrc)
-{
- GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
-
- GST_WASAPI2_SRC_LOCK (self);
-
- gst_clear_object (&self->client);
- gst_clear_caps (&self->cached_caps);
- self->started = FALSE;
-
- GST_WASAPI2_SRC_UNLOCK (self);
-
- return TRUE;
-}
-
-static gboolean
-gst_wasapi2_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
-{
- GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
- GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC (asrc);
- gboolean ret = FALSE;
- HRESULT hr;
-
- GST_WASAPI2_SRC_LOCK (self);
- if (!self->client && !gst_wasapi2_src_open_unlocked (asrc)) {
- GST_ERROR_OBJECT (self, "No audio client was configured");
- goto done;
- }
-
- if (!gst_wasapi2_client_ensure_activation (self->client)) {
- GST_ERROR_OBJECT (self, "Couldn't activate audio device");
- goto done;
- }
-
- hr = gst_wasapi2_client_open (self->client, spec, bsrc->ringbuffer);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (self, "Couldn't open audio client");
- goto done;
- }
-
- /* Set mute and volume here again, maybe when "mute" property was set, audioclient
- * might not be configured at that moment */
- if (self->mute_changed) {
- gst_wasapi2_client_set_mute (self->client, self->mute);
- self->mute_changed = FALSE;
- }
-
- if (self->volume_changed) {
- gst_wasapi2_client_set_volume (self->client, self->volume);
- self->volume_changed = FALSE;
- }
-
- /* Will start IAudioClient on the first read request */
- self->started = FALSE;
- ret = TRUE;
-
-done:
- GST_WASAPI2_SRC_UNLOCK (self);
-
- return ret;
-}
-
-static gboolean
-gst_wasapi2_src_unprepare (GstAudioSrc * asrc)
-{
- GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
-
- self->started = FALSE;
-
- /* Will reopen device later prepare() */
- GST_WASAPI2_SRC_LOCK (self);
- if (self->client) {
- gst_wasapi2_client_stop (self->client);
- gst_clear_object (&self->client);
- }
- GST_WASAPI2_SRC_UNLOCK (self);
-
- return TRUE;
-}
-
-static guint
-gst_wasapi2_src_read (GstAudioSrc * asrc, gpointer data, guint length,
- GstClockTime * timestamp)
+static GstAudioRingBuffer *
+gst_wasapi2_src_create_ringbuffer (GstAudioBaseSrc * src)
{
- GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
- HRESULT hr;
- guint read_len = 0;
+ GstWasapi2Src *self = GST_WASAPI2_SRC (src);
+ GstAudioRingBuffer *ringbuffer;
+ gchar *name;
- if (!self->client) {
- GST_ERROR_OBJECT (self, "No audio client was configured");
- return -1;
- }
-
- if (!self->started) {
- HRESULT hr = gst_wasapi2_client_start (self->client);
- if (!gst_wasapi2_result (hr)) {
- GST_ERROR_OBJECT (self, "Failed to re-start client");
- return -1;
- }
+ name = g_strdup_printf ("%s-ringbuffer", GST_OBJECT_NAME (src));
- self->started = TRUE;
- }
+ ringbuffer =
+ gst_wasapi2_ring_buffer_new (GST_WASAPI2_CLIENT_DEVICE_CLASS_CAPTURE,
+ self->low_latency, self->device_id, self->dispatcher, name);
+ g_free (name);
- hr = gst_wasapi2_client_read (self->client, data, length, &read_len);
- if (!gst_wasapi2_result (hr)) {
- GST_WARNING_OBJECT (self, "Failed to read data");
- return -1;
- }
-
- return read_len;
-}
-
-static guint
-gst_wasapi2_src_delay (GstAudioSrc * asrc)
-{
- GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
- guint32 delay;
- HRESULT hr;
-
- if (!self->client)
- return 0;
-
- hr = gst_wasapi2_client_delay (self->client, &delay);
- if (!gst_wasapi2_result (hr)) {
- GST_WARNING_OBJECT (self, "Failed to get delay");
- return 0;
- }
-
- return delay;
-}
-
-static void
-gst_wasapi2_src_reset (GstAudioSrc * asrc)
-{
- GstWasapi2Src *self = GST_WASAPI2_SRC (asrc);
-
- GST_DEBUG_OBJECT (self, "reset called");
-
- self->started = FALSE;
-
- if (!self->client)
- return;
-
- gst_wasapi2_client_stop (self->client);
+ return ringbuffer;
}
static void
gst_wasapi2_src_set_mute (GstWasapi2Src * self, gboolean mute)
{
- GST_WASAPI2_SRC_LOCK (self);
+ GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC_CAST (self);
+ HRESULT hr;
+
+ GST_OBJECT_LOCK (self);
self->mute = mute;
self->mute_changed = TRUE;
- if (self->client) {
- HRESULT hr = gst_wasapi2_client_set_mute (self->client, mute);
+ if (bsrc->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (bsrc->ringbuffer);
+
+ hr = gst_wasapi2_ring_buffer_set_mute (ringbuffer, mute);
if (FAILED (hr)) {
GST_INFO_OBJECT (self, "Couldn't set mute");
} else {
self->mute_changed = FALSE;
}
- } else {
- GST_DEBUG_OBJECT (self, "audio client is not configured yet");
}
- GST_WASAPI2_SRC_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
}
static gboolean
gst_wasapi2_src_get_mute (GstWasapi2Src * self)
{
+ GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC_CAST (self);
gboolean mute;
+ HRESULT hr;
- GST_WASAPI2_SRC_LOCK (self);
+ GST_OBJECT_LOCK (self);
mute = self->mute;
- if (self->client) {
- HRESULT hr = gst_wasapi2_client_get_mute (self->client, &mute);
+ if (bsrc->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (bsrc->ringbuffer);
+
+ hr = gst_wasapi2_ring_buffer_get_mute (ringbuffer, &mute);
+
if (FAILED (hr)) {
- GST_INFO_OBJECT (self, "Couldn't get mute state");
+ GST_INFO_OBJECT (self, "Couldn't get mute");
} else {
self->mute = mute;
}
- } else {
- GST_DEBUG_OBJECT (self, "audio client is not configured yet");
}
- GST_WASAPI2_SRC_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
return mute;
}
@@ -557,7 +393,10 @@ gst_wasapi2_src_get_mute (GstWasapi2Src * self)
static void
gst_wasapi2_src_set_volume (GstWasapi2Src * self, gdouble volume)
{
- GST_WASAPI2_SRC_LOCK (self);
+ GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC_CAST (self);
+ HRESULT hr;
+
+ GST_OBJECT_LOCK (self);
self->volume = volume;
/* clip volume value */
@@ -565,42 +404,47 @@ gst_wasapi2_src_set_volume (GstWasapi2Src * self, gdouble volume)
self->volume = MIN (1.0, self->volume);
self->volume_changed = TRUE;
- if (self->client) {
- HRESULT hr =
- gst_wasapi2_client_set_volume (self->client, (gfloat) self->volume);
+ if (bsrc->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (bsrc->ringbuffer);
+
+ hr = gst_wasapi2_ring_buffer_set_volume (ringbuffer, (gfloat) self->volume);
+
if (FAILED (hr)) {
GST_INFO_OBJECT (self, "Couldn't set volume");
} else {
self->volume_changed = FALSE;
}
- } else {
- GST_DEBUG_OBJECT (self, "audio client is not configured yet");
}
- GST_WASAPI2_SRC_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
}
static gdouble
gst_wasapi2_src_get_volume (GstWasapi2Src * self)
{
+ GstAudioBaseSrc *bsrc = GST_AUDIO_BASE_SRC_CAST (self);
gfloat volume;
+ HRESULT hr;
- GST_WASAPI2_SRC_LOCK (self);
+ GST_OBJECT_LOCK (self);
volume = (gfloat) self->volume;
- if (self->client) {
- HRESULT hr = gst_wasapi2_client_get_volume (self->client, &volume);
+ if (bsrc->ringbuffer) {
+ GstWasapi2RingBuffer *ringbuffer =
+ GST_WASAPI2_RING_BUFFER (bsrc->ringbuffer);
+
+ hr = gst_wasapi2_ring_buffer_get_volume (ringbuffer, &volume);
+
if (FAILED (hr)) {
- GST_INFO_OBJECT (self, "Couldn't get volume");
+ GST_INFO_OBJECT (self, "Couldn't set volume");
} else {
self->volume = volume;
}
- } else {
- GST_DEBUG_OBJECT (self, "audio client is not configured yet");
}
- GST_WASAPI2_SRC_UNLOCK (self);
+ GST_OBJECT_UNLOCK (self);
volume = MAX (0.0, volume);
volume = MIN (1.0, volume);
diff --git a/sys/wasapi2/gstwasapi2src.h b/sys/wasapi2/gstwasapi2src.h
index c832db5e8..2b3a0a0d6 100644
--- a/sys/wasapi2/gstwasapi2src.h
+++ b/sys/wasapi2/gstwasapi2src.h
@@ -27,7 +27,7 @@ G_BEGIN_DECLS
#define GST_TYPE_WASAPI2_SRC (gst_wasapi2_src_get_type ())
G_DECLARE_FINAL_TYPE (GstWasapi2Src,
- gst_wasapi2_src, GST, WASAPI2_SRC, GstAudioSrc);
+ gst_wasapi2_src, GST, WASAPI2_SRC, GstAudioBaseSrc);
G_END_DECLS
diff --git a/sys/wasapi2/gstwasapi2util.c b/sys/wasapi2/gstwasapi2util.c
index 81109d020..d19c96633 100644
--- a/sys/wasapi2/gstwasapi2util.c
+++ b/sys/wasapi2/gstwasapi2util.c
@@ -25,10 +25,56 @@
#include "gstwasapi2util.h"
#include <audioclient.h>
+#include <mmdeviceapi.h>
GST_DEBUG_CATEGORY_EXTERN (gst_wasapi2_debug);
#define GST_CAT_DEFAULT gst_wasapi2_debug
+/* Desktop only defines */
+#ifndef KSAUDIO_SPEAKER_MONO
+#define KSAUDIO_SPEAKER_MONO (SPEAKER_FRONT_CENTER)
+#endif
+#ifndef KSAUDIO_SPEAKER_1POINT1
+#define KSAUDIO_SPEAKER_1POINT1 (SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY)
+#endif
+#ifndef KSAUDIO_SPEAKER_STEREO
+#define KSAUDIO_SPEAKER_STEREO (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT)
+#endif
+#ifndef KSAUDIO_SPEAKER_2POINT1
+#define KSAUDIO_SPEAKER_2POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY)
+#endif
+#ifndef KSAUDIO_SPEAKER_3POINT0
+#define KSAUDIO_SPEAKER_3POINT0 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER)
+#endif
+#ifndef KSAUDIO_SPEAKER_3POINT1
+#define KSAUDIO_SPEAKER_3POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+ SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY)
+#endif
+#ifndef KSAUDIO_SPEAKER_QUAD
+#define KSAUDIO_SPEAKER_QUAD (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+ SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
+#endif
+#define KSAUDIO_SPEAKER_SURROUND (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+ SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER)
+#ifndef KSAUDIO_SPEAKER_5POINT0
+#define KSAUDIO_SPEAKER_5POINT0 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
+ SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
+#endif
+#define KSAUDIO_SPEAKER_5POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+ SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
+ SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
+#ifndef KSAUDIO_SPEAKER_7POINT0
+#define KSAUDIO_SPEAKER_7POINT0 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
+ SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
+ SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
+#endif
+#ifndef KSAUDIO_SPEAKER_7POINT1
+#define KSAUDIO_SPEAKER_7POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+ SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
+ SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
+ SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER)
+#endif
+
/* *INDENT-OFF* */
static struct
{
@@ -57,6 +103,27 @@ static struct
{SPEAKER_TOP_BACK_CENTER, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER},
{SPEAKER_TOP_BACK_RIGHT, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
};
+
+static DWORD default_ch_masks[] = {
+ 0,
+ KSAUDIO_SPEAKER_MONO,
+ /* 2ch */
+ KSAUDIO_SPEAKER_STEREO,
+ /* 2.1ch */
+ /* KSAUDIO_SPEAKER_3POINT0 ? */
+ KSAUDIO_SPEAKER_2POINT1,
+ /* 4ch */
+ /* KSAUDIO_SPEAKER_3POINT1 or KSAUDIO_SPEAKER_SURROUND ? */
+ KSAUDIO_SPEAKER_QUAD,
+ /* 5ch */
+ KSAUDIO_SPEAKER_5POINT0,
+ /* 5.1ch */
+ KSAUDIO_SPEAKER_5POINT1,
+ /* 7ch */
+ KSAUDIO_SPEAKER_7POINT0,
+ /* 7.1ch */
+ KSAUDIO_SPEAKER_7POINT1,
+};
/* *INDENT-ON* */
static const gchar *
@@ -192,3 +259,173 @@ _gst_wasapi2_result (HRESULT hr, GstDebugCategory * cat, const gchar * file,
return SUCCEEDED (hr);
#endif
}
+
+static void
+gst_wasapi_util_channel_position_all_none (guint channels,
+ GstAudioChannelPosition * position)
+{
+ guint i;
+
+ for (i = 0; i < channels; i++)
+ position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
+}
+
+guint64
+gst_wasapi2_util_waveformatex_to_channel_mask (WAVEFORMATEX * format,
+ GstAudioChannelPosition ** out_position)
+{
+ guint i, ch;
+ guint64 mask = 0;
+ GstAudioChannelPosition *pos = NULL;
+ WORD nChannels = 0;
+ DWORD dwChannelMask = 0;
+
+ nChannels = format->nChannels;
+ if (format->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
+ WAVEFORMATEXTENSIBLE *extensible = (WAVEFORMATEXTENSIBLE *) format;
+ dwChannelMask = extensible->dwChannelMask;
+ }
+
+ if (out_position)
+ *out_position = NULL;
+
+ if (nChannels > 2 && !dwChannelMask) {
+ GST_WARNING ("Unknown channel mask value for %d channel stream", nChannels);
+
+ if (nChannels >= G_N_ELEMENTS (default_ch_masks)) {
+ GST_ERROR ("To may channels %d", nChannels);
+ return 0;
+ }
+
+ dwChannelMask = default_ch_masks[nChannels];
+ }
+
+ pos = g_new (GstAudioChannelPosition, nChannels);
+ gst_wasapi_util_channel_position_all_none (nChannels, pos);
+
+ /* Too many channels, have to assume that they are all non-positional */
+ if (nChannels > G_N_ELEMENTS (wasapi_to_gst_pos)) {
+ GST_INFO ("Got too many (%i) channels, assuming non-positional", nChannels);
+ goto out;
+ }
+
+ /* Too many bits in the channel mask, and the bits don't match nChannels */
+ if (dwChannelMask >> (G_N_ELEMENTS (wasapi_to_gst_pos) + 1) != 0) {
+ GST_WARNING ("Too many bits in channel mask (%lu), assuming "
+ "non-positional", dwChannelMask);
+ goto out;
+ }
+
+ /* Map WASAPI's channel mask to Gstreamer's channel mask and positions.
+ * If the no. of bits in the mask > nChannels, we will ignore the extra. */
+ for (i = 0, ch = 0; i < G_N_ELEMENTS (wasapi_to_gst_pos) && ch < nChannels;
+ i++) {
+ if (!(dwChannelMask & wasapi_to_gst_pos[i].wasapi_pos))
+ /* no match, try next */
+ continue;
+ mask |= G_GUINT64_CONSTANT (1) << wasapi_to_gst_pos[i].gst_pos;
+ pos[ch++] = wasapi_to_gst_pos[i].gst_pos;
+ }
+
+ /* XXX: Warn if some channel masks couldn't be mapped? */
+
+ GST_DEBUG ("Converted WASAPI mask 0x%" G_GINT64_MODIFIER "x -> 0x%"
+ G_GINT64_MODIFIER "x", (guint64) dwChannelMask, (guint64) mask);
+
+out:
+ if (out_position) {
+ *out_position = pos;
+ } else {
+ g_free (pos);
+ }
+
+ return mask;
+}
+
+const gchar *
+gst_wasapi2_util_waveformatex_to_audio_format (WAVEFORMATEX * format)
+{
+ const gchar *fmt_str = NULL;
+ GstAudioFormat fmt = GST_AUDIO_FORMAT_UNKNOWN;
+
+ switch (format->wFormatTag) {
+ case WAVE_FORMAT_PCM:
+ fmt = gst_audio_format_build_integer (TRUE, G_LITTLE_ENDIAN,
+ format->wBitsPerSample, format->wBitsPerSample);
+ break;
+ case WAVE_FORMAT_IEEE_FLOAT:
+ if (format->wBitsPerSample == 32)
+ fmt = GST_AUDIO_FORMAT_F32LE;
+ else if (format->wBitsPerSample == 64)
+ fmt = GST_AUDIO_FORMAT_F64LE;
+ break;
+ case WAVE_FORMAT_EXTENSIBLE:
+ {
+ WAVEFORMATEXTENSIBLE *ex = (WAVEFORMATEXTENSIBLE *) format;
+ if (IsEqualGUID (&ex->SubFormat, &KSDATAFORMAT_SUBTYPE_PCM)) {
+ fmt = gst_audio_format_build_integer (TRUE, G_LITTLE_ENDIAN,
+ format->wBitsPerSample, ex->Samples.wValidBitsPerSample);
+ } else if (IsEqualGUID (&ex->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) {
+ if (format->wBitsPerSample == 32
+ && ex->Samples.wValidBitsPerSample == 32)
+ fmt = GST_AUDIO_FORMAT_F32LE;
+ else if (format->wBitsPerSample == 64 &&
+ ex->Samples.wValidBitsPerSample == 64)
+ fmt = GST_AUDIO_FORMAT_F64LE;
+ }
+ break;
+ }
+ default:
+ break;
+ }
+
+ if (fmt != GST_AUDIO_FORMAT_UNKNOWN)
+ fmt_str = gst_audio_format_to_string (fmt);
+
+ return fmt_str;
+}
+
+gboolean
+gst_wasapi2_util_parse_waveformatex (WAVEFORMATEX * format,
+ GstCaps * template_caps, GstCaps ** out_caps,
+ GstAudioChannelPosition ** out_positions)
+{
+ const gchar *afmt;
+ guint64 channel_mask;
+
+ *out_caps = NULL;
+
+ /* TODO: handle SPDIF and other encoded formats */
+
+ /* 1 or 2 channels <= 16 bits sample size OR
+ * 1 or 2 channels > 16 bits sample size or >2 channels */
+ if (format->wFormatTag != WAVE_FORMAT_PCM &&
+ format->wFormatTag != WAVE_FORMAT_IEEE_FLOAT &&
+ format->wFormatTag != WAVE_FORMAT_EXTENSIBLE)
+ /* Unhandled format tag */
+ return FALSE;
+
+ /* WASAPI can only tell us one canonical mix format that it will accept. The
+ * alternative is calling IsFormatSupported on all combinations of formats.
+ * Instead, it's simpler and faster to require conversion inside gstreamer */
+ afmt = gst_wasapi2_util_waveformatex_to_audio_format (format);
+ if (afmt == NULL)
+ return FALSE;
+
+ *out_caps = gst_caps_copy (template_caps);
+
+ channel_mask = gst_wasapi2_util_waveformatex_to_channel_mask (format,
+ out_positions);
+
+ gst_caps_set_simple (*out_caps,
+ "format", G_TYPE_STRING, afmt,
+ "channels", G_TYPE_INT, format->nChannels,
+ "rate", G_TYPE_INT, format->nSamplesPerSec, NULL);
+
+ if (channel_mask) {
+ gst_caps_set_simple (*out_caps,
+ "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
+ }
+
+ return TRUE;
+}
diff --git a/sys/wasapi2/gstwasapi2util.h b/sys/wasapi2/gstwasapi2util.h
index 67156f952..cb977fb94 100644
--- a/sys/wasapi2/gstwasapi2util.h
+++ b/sys/wasapi2/gstwasapi2util.h
@@ -23,6 +23,7 @@
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <windows.h>
+#include <initguid.h>
#include <audioclient.h>
G_BEGIN_DECLS
@@ -34,6 +35,13 @@ G_BEGIN_DECLS
"rate = " GST_AUDIO_RATE_RANGE ", " \
"channels = " GST_AUDIO_CHANNELS_RANGE
+#define GST_WASAPI2_CLEAR_COM(obj) G_STMT_START { \
+ if (obj) { \
+ (obj)->Release (); \
+ (obj) = NULL; \
+ } \
+ } G_STMT_END
+
gboolean _gst_wasapi2_result (HRESULT hr,
GstDebugCategory * cat,
const gchar * file,
@@ -43,6 +51,16 @@ gboolean _gst_wasapi2_result (HRESULT hr,
#define gst_wasapi2_result(result) \
_gst_wasapi2_result (result, GST_CAT_DEFAULT, __FILE__, GST_FUNCTION, __LINE__)
+guint64 gst_wasapi2_util_waveformatex_to_channel_mask (WAVEFORMATEX * format,
+ GstAudioChannelPosition ** out_position);
+
+const gchar * gst_wasapi2_util_waveformatex_to_audio_format (WAVEFORMATEX * format);
+
+gboolean gst_wasapi2_util_parse_waveformatex (WAVEFORMATEX * format,
+ GstCaps * template_caps,
+ GstCaps ** out_caps,
+ GstAudioChannelPosition ** out_positions);
+
G_END_DECLS
#endif /* __GST_WASAPI_UTIL_H__ */
diff --git a/sys/wasapi2/meson.build b/sys/wasapi2/meson.build
index 84d37c9ee..5edbfff2d 100644
--- a/sys/wasapi2/meson.build
+++ b/sys/wasapi2/meson.build
@@ -4,6 +4,7 @@ wasapi2_sources = [
'gstwasapi2util.c',
'gstwasapi2client.cpp',
'gstwasapi2device.c',
+ 'gstwasapi2ringbuffer.cpp',
'plugin.c',
]
@@ -26,7 +27,8 @@ ole32_dep = cc.find_library('ole32', required : get_option('wasapi2'))
ksuser_dep = cc.find_library('ksuser', required : get_option('wasapi2'))
runtimeobject_dep = cc.find_library('runtimeobject', required : get_option('wasapi2'))
mmdeviceapi_dep = cc.find_library('mmdevapi', required : get_option('wasapi2'))
-wasapi2_dep = [ole32_dep, ksuser_dep, runtimeobject_dep, mmdeviceapi_dep]
+mfplat_dep = cc.find_library('mfplat', required : get_option('wasapi2'))
+wasapi2_dep = [ole32_dep, ksuser_dep, runtimeobject_dep, mmdeviceapi_dep, mfplat_dep]
extra_args = []
foreach dep: wasapi2_dep
diff --git a/sys/wasapi2/plugin.c b/sys/wasapi2/plugin.c
index 18394ee0a..39cdf4c14 100644
--- a/sys/wasapi2/plugin.c
+++ b/sys/wasapi2/plugin.c
@@ -26,14 +26,23 @@
#include "gstwasapi2sink.h"
#include "gstwasapi2src.h"
#include "gstwasapi2device.h"
+#include "gstwasapi2util.h"
+#include <mfapi.h>
GST_DEBUG_CATEGORY (gst_wasapi2_debug);
GST_DEBUG_CATEGORY (gst_wasapi2_client_debug);
+static void
+plugin_deinit (gpointer data)
+{
+ MFShutdown ();
+}
+
static gboolean
plugin_init (GstPlugin * plugin)
{
GstRank rank = GST_RANK_SECONDARY;
+ HRESULT hr;
/**
* plugin-wasapi2:
@@ -41,6 +50,11 @@ plugin_init (GstPlugin * plugin)
* Since: 1.18
*/
+ hr = MFStartup (MF_VERSION, MFSTARTUP_NOSOCKET);
+ if (!gst_wasapi2_result (hr)) {
+ GST_WARNING ("MFStartup failure, hr: 0x%x", hr);
+ return TRUE;
+ }
#if WINAPI_FAMILY_PARTITION(WINAPI_PARTITION_APP) && !WINAPI_FAMILY_PARTITION(WINAPI_PARTITION_DESKTOP)
/* If we are building for UWP, wasapi2 plugin should have the highest rank */
rank = GST_RANK_PRIMARY + 1;
@@ -56,6 +70,10 @@ plugin_init (GstPlugin * plugin)
gst_device_provider_register (plugin, "wasapi2deviceprovider",
rank, GST_TYPE_WASAPI2_DEVICE_PROVIDER);
+ g_object_set_data_full (G_OBJECT (plugin),
+ "plugin-wasapi2-shutdown", "shutdown-data",
+ (GDestroyNotify) plugin_deinit);
+
return TRUE;
}
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