| /* |
| * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> |
| * Copyright (C) 2018 Centricular Ltd. |
| * Author: Nirbheek Chauhan <nirbheek@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. |
| */ |
| |
| /** |
| * SECTION:element-wasapisrc |
| * @title: wasapisrc |
| * |
| * Provides audio capture from the Windows Audio Session API available with |
| * Vista and newer. |
| * |
| * ## Example pipelines |
| * |[ |
| * gst-launch-1.0 -v wasapisrc ! fakesink |
| * ]| Capture from the default audio device and render to fakesink. |
| * |
| * |[ |
| * gst-launch-1.0 -v wasapisrc low-latency=true ! fakesink |
| * ]| Capture from the default audio device with the minimum possible latency and render to fakesink. |
| * |
| */ |
| #ifdef HAVE_CONFIG_H |
| # include <config.h> |
| #endif |
| |
| #include "gstwasapisrc.h" |
| |
| #include <avrt.h> |
| |
| GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug); |
| #define GST_CAT_DEFAULT gst_wasapi_src_debug |
| |
| static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS)); |
| |
| #define DEFAULT_ROLE GST_WASAPI_DEVICE_ROLE_CONSOLE |
| #define DEFAULT_EXCLUSIVE FALSE |
| #define DEFAULT_LOW_LATENCY FALSE |
| #define DEFAULT_AUDIOCLIENT3 FALSE |
| |
| enum |
| { |
| PROP_0, |
| PROP_ROLE, |
| PROP_DEVICE, |
| PROP_EXCLUSIVE, |
| PROP_LOW_LATENCY, |
| PROP_AUDIOCLIENT3 |
| }; |
| |
| static void gst_wasapi_src_dispose (GObject * object); |
| static void gst_wasapi_src_finalize (GObject * object); |
| static void gst_wasapi_src_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec); |
| static void gst_wasapi_src_get_property (GObject * object, guint prop_id, |
| GValue * value, GParamSpec * pspec); |
| |
| static GstCaps *gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter); |
| |
| static gboolean gst_wasapi_src_open (GstAudioSrc * asrc); |
| static gboolean gst_wasapi_src_close (GstAudioSrc * asrc); |
| static gboolean gst_wasapi_src_prepare (GstAudioSrc * asrc, |
| GstAudioRingBufferSpec * spec); |
| static gboolean gst_wasapi_src_unprepare (GstAudioSrc * asrc); |
| static guint gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, |
| guint length, GstClockTime * timestamp); |
| static guint gst_wasapi_src_delay (GstAudioSrc * asrc); |
| static void gst_wasapi_src_reset (GstAudioSrc * asrc); |
| |
| static GstClockTime gst_wasapi_src_get_time (GstClock * clock, |
| gpointer user_data); |
| |
| #define gst_wasapi_src_parent_class parent_class |
| G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_AUDIO_SRC); |
| |
| static void |
| gst_wasapi_src_class_init (GstWasapiSrcClass * klass) |
| { |
| GObjectClass *gobject_class = G_OBJECT_CLASS (klass); |
| GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); |
| GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass); |
| GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass); |
| |
| gobject_class->dispose = gst_wasapi_src_dispose; |
| gobject_class->finalize = gst_wasapi_src_finalize; |
| gobject_class->set_property = gst_wasapi_src_set_property; |
| gobject_class->get_property = gst_wasapi_src_get_property; |
| |
| g_object_class_install_property (gobject_class, |
| PROP_ROLE, |
| g_param_spec_enum ("role", "Role", |
| "Role of the device: communications, multimedia, etc", |
| GST_WASAPI_DEVICE_TYPE_ROLE, DEFAULT_ROLE, G_PARAM_READWRITE | |
| G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_READY)); |
| |
| g_object_class_install_property (gobject_class, |
| PROP_DEVICE, |
| g_param_spec_string ("device", "Device", |
| "WASAPI playback device as a GUID string", |
| NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| g_object_class_install_property (gobject_class, |
| PROP_EXCLUSIVE, |
| g_param_spec_boolean ("exclusive", "Exclusive mode", |
| "Open the device in exclusive mode", |
| DEFAULT_EXCLUSIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| 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, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| g_object_class_install_property (gobject_class, |
| PROP_AUDIOCLIENT3, |
| g_param_spec_boolean ("use-audioclient3", "Use the AudioClient3 API", |
| "Whether to use the Windows 10 AudioClient3 API when available", |
| DEFAULT_AUDIOCLIENT3, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| gst_element_class_add_static_pad_template (gstelement_class, &src_template); |
| gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc", |
| "Source/Audio", |
| "Stream audio from an audio capture device through WASAPI", |
| "Nirbheek Chauhan <nirbheek@centricular.com>, " |
| "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>"); |
| |
| gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_src_get_caps); |
| |
| gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_src_open); |
| gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_src_close); |
| gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_wasapi_src_read); |
| gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_prepare); |
| gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_unprepare); |
| gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_src_delay); |
| gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_src_reset); |
| |
| GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc", |
| 0, "Windows audio session API source"); |
| } |
| |
| static void |
| gst_wasapi_src_init (GstWasapiSrc * self) |
| { |
| /* override with a custom clock */ |
| if (GST_AUDIO_BASE_SRC (self)->clock) |
| gst_object_unref (GST_AUDIO_BASE_SRC (self)->clock); |
| |
| GST_AUDIO_BASE_SRC (self)->clock = gst_audio_clock_new ("GstWasapiSrcClock", |
| gst_wasapi_src_get_time, gst_object_ref (self), |
| (GDestroyNotify) gst_object_unref); |
| |
| self->role = DEFAULT_ROLE; |
| self->sharemode = AUDCLNT_SHAREMODE_SHARED; |
| self->low_latency = DEFAULT_LOW_LATENCY; |
| self->try_audioclient3 = DEFAULT_AUDIOCLIENT3; |
| self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL); |
| self->client_needs_restart = FALSE; |
| |
| CoInitialize (NULL); |
| } |
| |
| static void |
| gst_wasapi_src_dispose (GObject * object) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (object); |
| |
| if (self->event_handle != NULL) { |
| CloseHandle (self->event_handle); |
| self->event_handle = NULL; |
| } |
| |
| if (self->client_clock != NULL) { |
| IUnknown_Release (self->client_clock); |
| self->client_clock = NULL; |
| } |
| |
| if (self->client != NULL) { |
| IUnknown_Release (self->client); |
| self->client = NULL; |
| } |
| |
| if (self->capture_client != NULL) { |
| IUnknown_Release (self->capture_client); |
| self->capture_client = NULL; |
| } |
| |
| G_OBJECT_CLASS (parent_class)->dispose (object); |
| } |
| |
| static void |
| gst_wasapi_src_finalize (GObject * object) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (object); |
| |
| g_clear_pointer (&self->mix_format, CoTaskMemFree); |
| |
| g_clear_pointer (&self->cached_caps, gst_caps_unref); |
| g_clear_pointer (&self->positions, g_free); |
| g_clear_pointer (&self->device_strid, g_free); |
| |
| CoUninitialize (); |
| |
| G_OBJECT_CLASS (parent_class)->finalize (object); |
| } |
| |
| static void |
| gst_wasapi_src_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (object); |
| |
| switch (prop_id) { |
| case PROP_ROLE: |
| self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value)); |
| break; |
| case PROP_DEVICE: |
| { |
| const gchar *device = g_value_get_string (value); |
| g_free (self->device_strid); |
| self->device_strid = |
| device ? g_utf8_to_utf16 (device, -1, NULL, NULL, NULL) : NULL; |
| break; |
| } |
| case PROP_EXCLUSIVE: |
| self->sharemode = g_value_get_boolean (value) |
| ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED; |
| break; |
| case PROP_LOW_LATENCY: |
| self->low_latency = g_value_get_boolean (value); |
| break; |
| case PROP_AUDIOCLIENT3: |
| self->try_audioclient3 = g_value_get_boolean (value); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_wasapi_src_get_property (GObject * object, guint prop_id, |
| GValue * value, GParamSpec * pspec) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (object); |
| |
| switch (prop_id) { |
| case PROP_ROLE: |
| g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role)); |
| break; |
| case PROP_DEVICE: |
| g_value_take_string (value, self->device_strid ? |
| g_utf16_to_utf8 (self->device_strid, -1, NULL, NULL, NULL) : NULL); |
| break; |
| case PROP_EXCLUSIVE: |
| g_value_set_boolean (value, |
| self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE); |
| break; |
| case PROP_LOW_LATENCY: |
| g_value_set_boolean (value, self->low_latency); |
| break; |
| case PROP_AUDIOCLIENT3: |
| g_value_set_boolean (value, self->try_audioclient3); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static gboolean |
| gst_wasapi_src_can_audioclient3 (GstWasapiSrc * self) |
| { |
| if (self->sharemode == AUDCLNT_SHAREMODE_SHARED && |
| self->try_audioclient3 && gst_wasapi_util_have_audioclient3 ()) |
| return TRUE; |
| return FALSE; |
| } |
| |
| static GstCaps * |
| gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (bsrc); |
| WAVEFORMATEX *format = NULL; |
| GstCaps *caps = NULL; |
| |
| GST_DEBUG_OBJECT (self, "entering get caps"); |
| |
| if (self->cached_caps) { |
| caps = gst_caps_ref (self->cached_caps); |
| } else { |
| GstCaps *template_caps; |
| gboolean ret; |
| |
| template_caps = gst_pad_get_pad_template_caps (bsrc->srcpad); |
| |
| if (!self->client) { |
| caps = template_caps; |
| goto out; |
| } |
| |
| ret = gst_wasapi_util_get_device_format (GST_ELEMENT (self), |
| self->sharemode, self->device, self->client, &format); |
| if (!ret) { |
| GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), |
| ("failed to detect format")); |
| gst_caps_unref (template_caps); |
| return NULL; |
| } |
| |
| gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format, |
| template_caps, &caps, &self->positions); |
| if (caps == NULL) { |
| GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("unknown format")); |
| gst_caps_unref (template_caps); |
| return NULL; |
| } |
| |
| { |
| gchar *pos_str = gst_audio_channel_positions_to_string (self->positions, |
| format->nChannels); |
| GST_INFO_OBJECT (self, "positions are: %s", pos_str); |
| g_free (pos_str); |
| } |
| |
| self->mix_format = format; |
| gst_caps_replace (&self->cached_caps, caps); |
| gst_caps_unref (template_caps); |
| } |
| |
| if (filter) { |
| GstCaps *filtered = |
| gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); |
| gst_caps_unref (caps); |
| caps = filtered; |
| } |
| |
| out: |
| GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps); |
| return caps; |
| } |
| |
| static gboolean |
| gst_wasapi_src_open (GstAudioSrc * asrc) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| gboolean res = FALSE; |
| IAudioClient *client = NULL; |
| IMMDevice *device = NULL; |
| |
| if (self->client) |
| return TRUE; |
| |
| /* FIXME: Switching the default device does not switch the stream to it, |
| * even if the old device was unplugged. We need to handle this somehow. |
| * For example, perhaps we should automatically switch to the new device if |
| * the default device is changed and a device isn't explicitly selected. */ |
| if (!gst_wasapi_util_get_device_client (GST_ELEMENT (self), TRUE, |
| self->role, self->device_strid, &device, &client)) { |
| if (!self->device_strid) |
| GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL), |
| ("Failed to get default device")); |
| else |
| GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL), |
| ("Failed to open device %S", self->device_strid)); |
| goto beach; |
| } |
| |
| self->client = client; |
| self->device = device; |
| res = TRUE; |
| |
| beach: |
| |
| return res; |
| } |
| |
| static gboolean |
| gst_wasapi_src_close (GstAudioSrc * asrc) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| |
| if (self->device != NULL) { |
| IUnknown_Release (self->device); |
| self->device = NULL; |
| } |
| |
| if (self->client != NULL) { |
| IUnknown_Release (self->client); |
| self->client = NULL; |
| } |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| gboolean res = FALSE; |
| REFERENCE_TIME latency_rt; |
| guint bpf, rate, devicep_frames, buffer_frames; |
| HRESULT hr; |
| |
| CoInitialize (NULL); |
| |
| if (gst_wasapi_src_can_audioclient3 (self)) { |
| if (!gst_wasapi_util_initialize_audioclient3 (GST_ELEMENT (self), spec, |
| (IAudioClient3 *) self->client, self->mix_format, self->low_latency, |
| &devicep_frames)) |
| goto beach; |
| } else { |
| if (!gst_wasapi_util_initialize_audioclient (GST_ELEMENT (self), spec, |
| self->client, self->mix_format, self->sharemode, self->low_latency, |
| &devicep_frames)) |
| goto beach; |
| } |
| |
| 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 = IAudioClient_GetBufferSize (self->client, &buffer_frames); |
| HR_FAILED_GOTO (hr, IAudioClient::GetBufferSize, beach); |
| |
| GST_INFO_OBJECT (self, "buffer size is %i frames, device period is %i " |
| "frames, bpf is %i bytes, rate is %i Hz", buffer_frames, |
| devicep_frames, bpf, rate); |
| |
| /* Actual latency-time/buffer-time will be different now */ |
| spec->segsize = devicep_frames * bpf; |
| |
| /* We need a minimum of 2 segments to ensure glitch-free playback */ |
| spec->segtotal = MAX (self->buffer_frame_count * bpf / spec->segsize, 2); |
| |
| GST_INFO_OBJECT (self, "segsize is %i, segtotal is %i", spec->segsize, |
| spec->segtotal); |
| |
| /* Get WASAPI latency for logging */ |
| hr = IAudioClient_GetStreamLatency (self->client, &latency_rt); |
| HR_FAILED_GOTO (hr, IAudioClient::GetStreamLatency, beach); |
| |
| GST_INFO_OBJECT (self, "wasapi stream latency: %" G_GINT64_FORMAT " (%" |
| G_GINT64_FORMAT " ms)", latency_rt, latency_rt / 10000); |
| |
| /* Set the event handler which will trigger reads */ |
| hr = IAudioClient_SetEventHandle (self->client, self->event_handle); |
| HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach); |
| |
| /* Get the clock and the clock freq */ |
| if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client, |
| &self->client_clock)) |
| goto beach; |
| |
| hr = IAudioClock_GetFrequency (self->client_clock, &self->client_clock_freq); |
| HR_FAILED_GOTO (hr, IAudioClock::GetFrequency, beach); |
| |
| /* Get capture source client and start it up */ |
| if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client, |
| &self->capture_client)) { |
| goto beach; |
| } |
| |
| hr = IAudioClient_Start (self->client); |
| HR_FAILED_GOTO (hr, IAudioClock::Start, beach); |
| |
| gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC |
| (self)->ringbuffer, self->positions); |
| |
| /* Increase the thread priority to reduce glitches */ |
| self->thread_priority_handle = gst_wasapi_util_set_thread_characteristics (); |
| |
| res = TRUE; |
| beach: |
| /* unprepare() is not called if prepare() fails, but we want it to be, so call |
| * it manually when needed */ |
| if (!res) |
| gst_wasapi_src_unprepare (asrc); |
| |
| return res; |
| } |
| |
| static gboolean |
| gst_wasapi_src_unprepare (GstAudioSrc * asrc) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| |
| CoUninitialize (); |
| |
| if (self->thread_priority_handle != NULL) { |
| gst_wasapi_util_revert_thread_characteristics |
| (self->thread_priority_handle); |
| self->thread_priority_handle = NULL; |
| } |
| |
| if (self->client != NULL) { |
| IAudioClient_Stop (self->client); |
| } |
| |
| if (self->capture_client != NULL) { |
| IUnknown_Release (self->capture_client); |
| self->capture_client = NULL; |
| } |
| |
| if (self->client_clock != NULL) { |
| IUnknown_Release (self->client_clock); |
| self->client_clock = NULL; |
| } |
| |
| self->client_clock_freq = 0; |
| |
| return TRUE; |
| } |
| |
| static guint |
| gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length, |
| GstClockTime * timestamp) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| HRESULT hr; |
| gint16 *from = NULL; |
| guint wanted = length; |
| DWORD flags; |
| |
| GST_OBJECT_LOCK (self); |
| if (self->client_needs_restart) { |
| hr = IAudioClient_Start (self->client); |
| HR_FAILED_AND (hr, IAudioClient::Start, length = 0; goto beach); |
| self->client_needs_restart = FALSE; |
| } |
| GST_OBJECT_UNLOCK (self); |
| |
| while (wanted > 0) { |
| DWORD dwWaitResult; |
| guint have_frames, n_frames, want_frames, read_len; |
| |
| /* Wait for data to become available */ |
| dwWaitResult = WaitForSingleObject (self->event_handle, INFINITE); |
| if (dwWaitResult != WAIT_OBJECT_0) { |
| GST_ERROR_OBJECT (self, "Error waiting for event handle: %x", |
| (guint) dwWaitResult); |
| length = 0; |
| goto beach; |
| } |
| |
| hr = IAudioCaptureClient_GetBuffer (self->capture_client, |
| (BYTE **) & from, &have_frames, &flags, NULL, NULL); |
| if (hr != S_OK) { |
| gchar *msg = gst_wasapi_util_hresult_to_string (hr); |
| if (hr == AUDCLNT_S_BUFFER_EMPTY) |
| GST_WARNING_OBJECT (self, "IAudioCaptureClient::GetBuffer failed: %s" |
| ", retrying", msg); |
| else |
| GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer failed: %s", |
| msg); |
| g_free (msg); |
| length = 0; |
| goto beach; |
| } |
| |
| if (flags != 0) |
| GST_INFO_OBJECT (self, "buffer flags=%#08x", (guint) flags); |
| |
| /* XXX: How do we handle AUDCLNT_BUFFERFLAGS_SILENT? We're supposed to write |
| * out silence when that flag is set? See: |
| * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370800(v=vs.85).aspx */ |
| |
| if (flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) |
| GST_WARNING_OBJECT (self, "WASAPI reported glitch in buffer"); |
| |
| want_frames = wanted / self->mix_format->nBlockAlign; |
| |
| /* If GetBuffer is returning more frames than we can handle, all we can do is |
| * hope that this is temporary and that things will settle down later. */ |
| if (G_UNLIKELY (have_frames > want_frames)) |
| GST_WARNING_OBJECT (self, "captured too many frames: have %i, want %i", |
| have_frames, want_frames); |
| |
| /* Only copy data that will fit into the allocated buffer of size @length */ |
| n_frames = MIN (have_frames, want_frames); |
| read_len = n_frames * self->mix_format->nBlockAlign; |
| |
| { |
| guint bpf = self->mix_format->nBlockAlign; |
| GST_DEBUG_OBJECT (self, "have: %i (%i bytes), can read: %i (%i bytes), " |
| "will read: %i (%i bytes)", have_frames, have_frames * bpf, |
| want_frames, wanted, n_frames, read_len); |
| } |
| |
| memcpy (data, from, read_len); |
| wanted -= read_len; |
| |
| /* Always release all captured buffers if we've captured any at all */ |
| hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, have_frames); |
| HR_FAILED_AND (hr, IAudioClock::ReleaseBuffer, goto beach); |
| } |
| |
| |
| beach: |
| |
| return length; |
| } |
| |
| static guint |
| gst_wasapi_src_delay (GstAudioSrc * asrc) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| guint delay = 0; |
| HRESULT hr; |
| |
| hr = IAudioClient_GetCurrentPadding (self->client, &delay); |
| HR_FAILED_RET (hr, IAudioClock::GetCurrentPadding, 0); |
| |
| return delay; |
| } |
| |
| static void |
| gst_wasapi_src_reset (GstAudioSrc * asrc) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| HRESULT hr; |
| |
| if (!self->client) |
| return; |
| |
| GST_OBJECT_LOCK (self); |
| hr = IAudioClient_Stop (self->client); |
| HR_FAILED_RET (hr, IAudioClock::Stop,); |
| |
| hr = IAudioClient_Reset (self->client); |
| HR_FAILED_RET (hr, IAudioClock::Reset,); |
| |
| self->client_needs_restart = TRUE; |
| GST_OBJECT_UNLOCK (self); |
| } |
| |
| static GstClockTime |
| gst_wasapi_src_get_time (GstClock * clock, gpointer user_data) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (user_data); |
| HRESULT hr; |
| guint64 devpos; |
| GstClockTime result; |
| |
| if (G_UNLIKELY (self->client_clock == NULL)) |
| return GST_CLOCK_TIME_NONE; |
| |
| hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL); |
| HR_FAILED_RET (hr, IAudioClock::GetPosition, GST_CLOCK_TIME_NONE); |
| |
| result = gst_util_uint64_scale_int (devpos, GST_SECOND, |
| self->client_clock_freq); |
| |
| /* |
| GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT |
| " frequency = %" G_GUINT64_FORMAT |
| " result = %" G_GUINT64_FORMAT " ms", |
| devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result)); |
| */ |
| |
| return result; |
| } |