/* * Copyright (C) 2008 Ole André Vadla Ravnås * Copyright (C) 2018 Centricular Ltd. * Author: Nirbheek Chauhan * * 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. */ #ifdef HAVE_CONFIG_H # include #endif /* Note: initguid.h can not be included in gstwasapiutil.h, otherwise a * symbol redefinition error will be raised. * initguid.h must be included in the C file before mmdeviceapi.h * which is included in gstwasapiutil.h. */ #ifdef _MSC_VER #include #endif #include "gstwasapiutil.h" #include "gstwasapidevice.h" GST_DEBUG_CATEGORY_EXTERN (gst_wasapi_debug); #define GST_CAT_DEFAULT gst_wasapi_debug /* This was only added to MinGW in ~2015 and our Cerbero toolchain is too old */ #if defined(_MSC_VER) #include #elif !defined(PKEY_Device_FriendlyName) #include #include DEFINE_PROPERTYKEY (PKEY_Device_FriendlyName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 14); DEFINE_PROPERTYKEY (PKEY_AudioEngine_DeviceFormat, 0xf19f064d, 0x82c, 0x4e27, 0xbc, 0x73, 0x68, 0x82, 0xa1, 0xbb, 0x8e, 0x4c, 0); #endif /* __uuidof is only available in C++, so we hard-code the GUID values for all * these. This is ok because these are ABI. */ const CLSID CLSID_MMDeviceEnumerator = { 0xbcde0395, 0xe52f, 0x467c, {0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e} }; const IID IID_IMMDeviceEnumerator = { 0xa95664d2, 0x9614, 0x4f35, {0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6} }; const IID IID_IMMEndpoint = { 0x1be09788, 0x6894, 0x4089, {0x85, 0x86, 0x9a, 0x2a, 0x6c, 0x26, 0x5a, 0xc5} }; const IID IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32, {0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2} }; const IID IID_IAudioClient3 = { 0x7ed4ee07, 0x8e67, 0x4cd4, {0x8c, 0x1a, 0x2b, 0x7a, 0x59, 0x87, 0xad, 0x42} }; const IID IID_IAudioClock = { 0xcd63314f, 0x3fba, 0x4a1b, {0x81, 0x2c, 0xef, 0x96, 0x35, 0x87, 0x28, 0xe7} }; const IID IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0, {0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17} }; const IID IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483, {0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2} }; /* *INDENT-OFF* */ 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} }; /* *INDENT-ON* */ static int windows_major_version = 0; gboolean gst_wasapi_util_have_audioclient3 (void) { if (windows_major_version > 0) return windows_major_version == 10; if (g_getenv ("GST_WASAPI_DISABLE_AUDIOCLIENT3") != NULL) { windows_major_version = 6; return FALSE; } /* https://msdn.microsoft.com/en-us/library/windows/desktop/ms724834(v=vs.85).aspx */ windows_major_version = 6; if (g_win32_check_windows_version (10, 0, 0, G_WIN32_OS_ANY)) windows_major_version = 10; return windows_major_version == 10; } GType gst_wasapi_device_role_get_type (void) { static const GEnumValue values[] = { {GST_WASAPI_DEVICE_ROLE_CONSOLE, "Games, system notifications, voice commands", "console"}, {GST_WASAPI_DEVICE_ROLE_MULTIMEDIA, "Music, movies, recorded media", "multimedia"}, {GST_WASAPI_DEVICE_ROLE_COMMS, "Voice communications", "comms"}, {0, NULL, NULL} }; static volatile GType id = 0; if (g_once_init_enter ((gsize *) & id)) { GType _id; _id = g_enum_register_static ("GstWasapiDeviceRole", values); g_once_init_leave ((gsize *) & id, _id); } return id; } gint gst_wasapi_device_role_to_erole (gint role) { switch (role) { case GST_WASAPI_DEVICE_ROLE_CONSOLE: return eConsole; case GST_WASAPI_DEVICE_ROLE_MULTIMEDIA: return eMultimedia; case GST_WASAPI_DEVICE_ROLE_COMMS: return eCommunications; default: g_assert_not_reached (); } } gint gst_wasapi_erole_to_device_role (gint erole) { switch (erole) { case eConsole: return GST_WASAPI_DEVICE_ROLE_CONSOLE; case eMultimedia: return GST_WASAPI_DEVICE_ROLE_MULTIMEDIA; case eCommunications: return GST_WASAPI_DEVICE_ROLE_COMMS; default: g_assert_not_reached (); } } static const gchar * hresult_to_string_fallback (HRESULT hr) { const gchar *s = "unknown error"; switch (hr) { case AUDCLNT_E_NOT_INITIALIZED: s = "AUDCLNT_E_NOT_INITIALIZED"; break; case AUDCLNT_E_ALREADY_INITIALIZED: s = "AUDCLNT_E_ALREADY_INITIALIZED"; break; case AUDCLNT_E_WRONG_ENDPOINT_TYPE: s = "AUDCLNT_E_WRONG_ENDPOINT_TYPE"; break; case AUDCLNT_E_DEVICE_INVALIDATED: s = "AUDCLNT_E_DEVICE_INVALIDATED"; break; case AUDCLNT_E_NOT_STOPPED: s = "AUDCLNT_E_NOT_STOPPED"; break; case AUDCLNT_E_BUFFER_TOO_LARGE: s = "AUDCLNT_E_BUFFER_TOO_LARGE"; break; case AUDCLNT_E_OUT_OF_ORDER: s = "AUDCLNT_E_OUT_OF_ORDER"; break; case AUDCLNT_E_UNSUPPORTED_FORMAT: s = "AUDCLNT_E_UNSUPPORTED_FORMAT"; break; case AUDCLNT_E_INVALID_DEVICE_PERIOD: s = "AUDCLNT_E_INVALID_DEVICE_PERIOD"; break; case AUDCLNT_E_INVALID_SIZE: s = "AUDCLNT_E_INVALID_SIZE"; break; case AUDCLNT_E_DEVICE_IN_USE: s = "AUDCLNT_E_DEVICE_IN_USE"; break; case AUDCLNT_E_BUFFER_OPERATION_PENDING: s = "AUDCLNT_E_BUFFER_OPERATION_PENDING"; break; case AUDCLNT_E_BUFFER_SIZE_ERROR: s = "AUDCLNT_E_BUFFER_SIZE_ERROR"; break; case AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED: s = "AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED"; break; case AUDCLNT_E_THREAD_NOT_REGISTERED: s = "AUDCLNT_E_THREAD_NOT_REGISTERED"; break; case AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED: s = "AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED"; break; case AUDCLNT_E_ENDPOINT_CREATE_FAILED: s = "AUDCLNT_E_ENDPOINT_CREATE_FAILED"; break; case AUDCLNT_E_SERVICE_NOT_RUNNING: s = "AUDCLNT_E_SERVICE_NOT_RUNNING"; break; case AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED: s = "AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED"; break; case AUDCLNT_E_EXCLUSIVE_MODE_ONLY: s = "AUDCLNT_E_EXCLUSIVE_MODE_ONLY"; break; case AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL: s = "AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL"; break; case AUDCLNT_E_EVENTHANDLE_NOT_SET: s = "AUDCLNT_E_EVENTHANDLE_NOT_SET"; break; case AUDCLNT_E_INCORRECT_BUFFER_SIZE: s = "AUDCLNT_E_INCORRECT_BUFFER_SIZE"; break; case AUDCLNT_E_CPUUSAGE_EXCEEDED: s = "AUDCLNT_E_CPUUSAGE_EXCEEDED"; break; case AUDCLNT_S_BUFFER_EMPTY: s = "AUDCLNT_S_BUFFER_EMPTY"; break; case AUDCLNT_S_THREAD_ALREADY_REGISTERED: s = "AUDCLNT_S_THREAD_ALREADY_REGISTERED"; break; case AUDCLNT_S_POSITION_STALLED: s = "AUDCLNT_S_POSITION_STALLED"; break; case E_POINTER: s = "E_POINTER"; break; case E_INVALIDARG: s = "E_INVALIDARG"; break; } return s; } gchar * gst_wasapi_util_hresult_to_string (HRESULT hr) { DWORD flags; gchar *ret_text; LPTSTR error_text = NULL; flags = FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_ALLOCATE_BUFFER | FORMAT_MESSAGE_IGNORE_INSERTS; FormatMessage (flags, NULL, hr, MAKELANGID (LANG_NEUTRAL, SUBLANG_DEFAULT), (LPTSTR) & error_text, 0, NULL); /* If we couldn't get the error msg, try the fallback switch statement */ if (error_text == NULL) return g_strdup (hresult_to_string_fallback (hr)); #ifdef UNICODE /* If UNICODE is defined, LPTSTR is LPWSTR which is UTF-16 */ ret_text = g_utf16_to_utf8 (error_text, 0, NULL, NULL, NULL); #else ret_text = g_strdup (error_text); #endif LocalFree (error_text); return ret_text; } static IMMDeviceEnumerator * gst_wasapi_util_get_device_enumerator (GstElement * self) { HRESULT hr; IMMDeviceEnumerator *enumerator = NULL; hr = CoCreateInstance (&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, &IID_IMMDeviceEnumerator, (void **) &enumerator); HR_FAILED_RET (hr, CoCreateInstance (MMDeviceEnumerator), NULL); return enumerator; } gboolean gst_wasapi_util_get_devices (GstElement * self, gboolean active, GList ** devices) { gboolean res = FALSE; static GstStaticCaps scaps = GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS); DWORD dwStateMask = active ? DEVICE_STATE_ACTIVE : DEVICE_STATEMASK_ALL; IMMDeviceCollection *device_collection = NULL; IMMDeviceEnumerator *enumerator = NULL; const gchar *device_class, *element_name; guint ii, count; HRESULT hr; *devices = NULL; enumerator = gst_wasapi_util_get_device_enumerator (self); if (!enumerator) return FALSE; hr = IMMDeviceEnumerator_EnumAudioEndpoints (enumerator, eAll, dwStateMask, &device_collection); HR_FAILED_GOTO (hr, IMMDeviceEnumerator::EnumAudioEndpoints, err); hr = IMMDeviceCollection_GetCount (device_collection, &count); HR_FAILED_GOTO (hr, IMMDeviceCollection::GetCount, err); /* Create a GList of GstDevices* to return */ for (ii = 0; ii < count; ii++) { IMMDevice *item = NULL; IMMEndpoint *endpoint = NULL; IAudioClient *client = NULL; IPropertyStore *prop_store = NULL; WAVEFORMATEX *format = NULL; gchar *description = NULL; gchar *strid = NULL; EDataFlow dataflow; PROPVARIANT var; wchar_t *wstrid; GstDevice *device; GstStructure *props; GstCaps *caps; hr = IMMDeviceCollection_Item (device_collection, ii, &item); if (hr != S_OK) continue; hr = IMMDevice_QueryInterface (item, &IID_IMMEndpoint, (void **) &endpoint); if (hr != S_OK) goto next; hr = IMMEndpoint_GetDataFlow (endpoint, &dataflow); if (hr != S_OK) goto next; if (dataflow == eRender) { device_class = "Audio/Sink"; element_name = "wasapisink"; } else { device_class = "Audio/Source"; element_name = "wasapisrc"; } PropVariantInit (&var); hr = IMMDevice_GetId (item, &wstrid); if (hr != S_OK) goto next; strid = g_utf16_to_utf8 (wstrid, -1, NULL, NULL, NULL); CoTaskMemFree (wstrid); hr = IMMDevice_OpenPropertyStore (item, STGM_READ, &prop_store); if (hr != S_OK) goto next; /* NOTE: More properties can be added as needed from here: * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370794(v=vs.85).aspx */ hr = IPropertyStore_GetValue (prop_store, &PKEY_Device_FriendlyName, &var); if (hr != S_OK) goto next; description = g_utf16_to_utf8 (var.pwszVal, -1, NULL, NULL, NULL); PropVariantClear (&var); /* Get the audio client so we can fetch the mix format for shared mode * to get the device format for exclusive mode (or something close to that) * fetch PKEY_AudioEngine_DeviceFormat from the property store. */ hr = IMMDevice_Activate (item, &IID_IAudioClient, CLSCTX_ALL, NULL, (void **) &client); if (hr != S_OK) { gchar *msg = gst_wasapi_util_hresult_to_string (hr); GST_ERROR_OBJECT (self, "IMMDevice::Activate (IID_IAudioClient) failed" "on %s: %s", strid, msg); g_free (msg); goto next; } hr = IAudioClient_GetMixFormat (client, &format); if (hr != S_OK || format == NULL) { gchar *msg = gst_wasapi_util_hresult_to_string (hr); GST_ERROR_OBJECT (self, "GetMixFormat failed on %s: %s", strid, msg); g_free (msg); goto next; } if (!gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format, gst_static_caps_get (&scaps), &caps, NULL)) goto next; /* Set some useful properties */ props = gst_structure_new ("wasapi-proplist", "device.api", G_TYPE_STRING, "wasapi", "device.strid", G_TYPE_STRING, GST_STR_NULL (strid), "wasapi.device.description", G_TYPE_STRING, description, NULL); device = g_object_new (GST_TYPE_WASAPI_DEVICE, "device", strid, "display-name", description, "caps", caps, "device-class", device_class, "properties", props, NULL); GST_WASAPI_DEVICE (device)->element = element_name; gst_structure_free (props); gst_caps_unref (caps); *devices = g_list_prepend (*devices, device); next: PropVariantClear (&var); if (prop_store) IUnknown_Release (prop_store); if (endpoint) IUnknown_Release (endpoint); if (client) IUnknown_Release (client); if (item) IUnknown_Release (item); if (description) g_free (description); if (strid) g_free (strid); } res = TRUE; err: if (enumerator) IUnknown_Release (enumerator); if (device_collection) IUnknown_Release (device_collection); return res; } gboolean gst_wasapi_util_get_device_format (GstElement * self, gint device_mode, IMMDevice * device, IAudioClient * client, WAVEFORMATEX ** ret_format) { WAVEFORMATEX *format; HRESULT hr; *ret_format = NULL; hr = IAudioClient_GetMixFormat (client, &format); HR_FAILED_RET (hr, IAudioClient::GetMixFormat, FALSE); /* WASAPI always accepts the format returned by GetMixFormat in shared mode */ if (device_mode == AUDCLNT_SHAREMODE_SHARED) goto out; /* WASAPI may or may not support this format in exclusive mode */ hr = IAudioClient_IsFormatSupported (client, AUDCLNT_SHAREMODE_EXCLUSIVE, format, NULL); if (hr == S_OK) goto out; CoTaskMemFree (format); /* Open the device property store, and get the format that WASAPI has been * using for sending data to the device */ { PROPVARIANT var; IPropertyStore *prop_store = NULL; hr = IMMDevice_OpenPropertyStore (device, STGM_READ, &prop_store); HR_FAILED_RET (hr, IMMDevice::OpenPropertyStore, FALSE); hr = IPropertyStore_GetValue (prop_store, &PKEY_AudioEngine_DeviceFormat, &var); if (hr != S_OK) { gchar *msg = gst_wasapi_util_hresult_to_string (hr); GST_ERROR_OBJECT (self, "GetValue failed: %s", msg); g_free (msg); IUnknown_Release (prop_store); return FALSE; } format = malloc (var.blob.cbSize); memcpy (format, var.blob.pBlobData, var.blob.cbSize); PropVariantClear (&var); IUnknown_Release (prop_store); } /* WASAPI may or may not support this format in exclusive mode */ hr = IAudioClient_IsFormatSupported (client, AUDCLNT_SHAREMODE_EXCLUSIVE, format, NULL); if (hr == S_OK) goto out; GST_ERROR_OBJECT (self, "AudioEngine DeviceFormat not supported"); free (format); return FALSE; out: *ret_format = format; return TRUE; } gboolean gst_wasapi_util_get_device_client (GstElement * self, gint data_flow, gint role, const wchar_t * device_strid, IMMDevice ** ret_device, IAudioClient ** ret_client) { gboolean res = FALSE; HRESULT hr; IMMDeviceEnumerator *enumerator = NULL; IMMDevice *device = NULL; IAudioClient *client = NULL; if (!(enumerator = gst_wasapi_util_get_device_enumerator (self))) goto beach; if (!device_strid) { hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint (enumerator, data_flow, role, &device); HR_FAILED_GOTO (hr, IMMDeviceEnumerator::GetDefaultAudioEndpoint, beach); } else { hr = IMMDeviceEnumerator_GetDevice (enumerator, device_strid, &device); if (hr != S_OK) { gchar *msg = gst_wasapi_util_hresult_to_string (hr); GST_ERROR_OBJECT (self, "IMMDeviceEnumerator::GetDevice (%S) failed" ": %s", device_strid, msg); g_free (msg); goto beach; } } if (gst_wasapi_util_have_audioclient3 ()) hr = IMMDevice_Activate (device, &IID_IAudioClient3, CLSCTX_ALL, NULL, (void **) &client); else hr = IMMDevice_Activate (device, &IID_IAudioClient, CLSCTX_ALL, NULL, (void **) &client); HR_FAILED_GOTO (hr, IMMDevice::Activate (IID_IAudioClient), beach); IUnknown_AddRef (client); IUnknown_AddRef (device); *ret_client = client; *ret_device = device; res = TRUE; beach: if (client != NULL) IUnknown_Release (client); if (device != NULL) IUnknown_Release (device); if (enumerator != NULL) IUnknown_Release (enumerator); return res; } gboolean gst_wasapi_util_get_render_client (GstElement * self, IAudioClient * client, IAudioRenderClient ** ret_render_client) { gboolean res = FALSE; HRESULT hr; IAudioRenderClient *render_client = NULL; hr = IAudioClient_GetService (client, &IID_IAudioRenderClient, (void **) &render_client); HR_FAILED_GOTO (hr, IAudioClient::GetService, beach); *ret_render_client = render_client; res = TRUE; beach: return res; } gboolean gst_wasapi_util_get_capture_client (GstElement * self, IAudioClient * client, IAudioCaptureClient ** ret_capture_client) { gboolean res = FALSE; HRESULT hr; IAudioCaptureClient *capture_client = NULL; hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient, (void **) &capture_client); HR_FAILED_GOTO (hr, IAudioClient::GetService, beach); *ret_capture_client = capture_client; res = TRUE; beach: return res; } gboolean gst_wasapi_util_get_clock (GstElement * self, IAudioClient * client, IAudioClock ** ret_clock) { gboolean res = FALSE; HRESULT hr; IAudioClock *clock = NULL; hr = IAudioClient_GetService (client, &IID_IAudioClock, (void **) &clock); HR_FAILED_GOTO (hr, IAudioClient::GetService, beach); *ret_clock = clock; res = TRUE; beach: return res; } 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; } /* 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_wasapi_util_waveformatex_to_channel_mask (WAVEFORMATEXTENSIBLE * format, GstAudioChannelPosition ** out_position) { int ii, ch; guint64 mask = 0; WORD nChannels = format->Format.nChannels; DWORD dwChannelMask = format->dwChannelMask; 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; } gboolean gst_wasapi_util_parse_waveformatex (WAVEFORMATEXTENSIBLE * format, GstCaps * template_caps, GstCaps ** out_caps, GstAudioChannelPosition ** out_positions) { int ii; 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->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); /* This will always return something that might be usable */ channel_mask = gst_wasapi_util_waveformatex_to_channel_mask (format, out_positions); for (ii = 0; ii < gst_caps_get_size (*out_caps); ii++) { GstStructure *s = gst_caps_get_structure (*out_caps, ii); gst_structure_set (s, "format", G_TYPE_STRING, afmt, "channels", G_TYPE_INT, format->Format.nChannels, "rate", G_TYPE_INT, format->Format.nSamplesPerSec, NULL); if (channel_mask) { gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL); } } return TRUE; } void gst_wasapi_util_get_best_buffer_sizes (GstAudioRingBufferSpec * spec, gboolean exclusive, REFERENCE_TIME default_period, REFERENCE_TIME min_period, REFERENCE_TIME * ret_period, REFERENCE_TIME * ret_buffer_duration) { REFERENCE_TIME use_period, use_buffer; /* Figure out what integral device period to use as the base */ if (exclusive) { /* Exclusive mode can run at multiples of either the minimum period or the * default period; these are on the hardware ringbuffer */ if (spec->latency_time * 10 > default_period) use_period = default_period; else use_period = min_period; } else { /* 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); if (exclusive) { /* Buffer duration is the same as the period in exclusive mode. The * hardware is always writing out one buffer (of size *ret_period), and * we're writing to the other one. */ use_buffer = use_period; } else { /* 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; } gboolean gst_wasapi_util_initialize_audioclient (GstElement * self, GstAudioRingBufferSpec * spec, IAudioClient * client, WAVEFORMATEX * format, guint sharemode, gboolean low_latency, gboolean loopback, guint * ret_devicep_frames) { REFERENCE_TIME default_period, min_period; REFERENCE_TIME device_period, device_buffer_duration; guint rate, stream_flags; guint32 n_frames; HRESULT hr; hr = IAudioClient_GetDevicePeriod (client, &default_period, &min_period); HR_FAILED_RET (hr, IAudioClient::GetDevicePeriod, FALSE); GST_INFO_OBJECT (self, "wasapi default period: %" G_GINT64_FORMAT ", min period: %" G_GINT64_FORMAT, default_period, min_period); rate = GST_AUDIO_INFO_RATE (&spec->info); if (low_latency) { if (sharemode == AUDCLNT_SHAREMODE_SHARED) { device_period = default_period; device_buffer_duration = 0; } else { device_period = min_period; device_buffer_duration = min_period; } } else { /* Clamp values to integral multiples of an appropriate period */ gst_wasapi_util_get_best_buffer_sizes (spec, sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE, default_period, min_period, &device_period, &device_buffer_duration); } stream_flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK; if (loopback) stream_flags |= AUDCLNT_STREAMFLAGS_LOOPBACK; hr = IAudioClient_Initialize (client, sharemode, stream_flags, device_buffer_duration, /* This must always be 0 in shared mode */ sharemode == AUDCLNT_SHAREMODE_SHARED ? 0 : device_period, format, NULL); if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED && sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) { GST_WARNING_OBJECT (self, "initialize failed due to unaligned period %i", (int) device_period); /* Calculate a new aligned period. First get the aligned buffer size. */ hr = IAudioClient_GetBufferSize (client, &n_frames); HR_FAILED_RET (hr, IAudioClient::GetBufferSize, FALSE); device_period = (GST_SECOND / 100) * n_frames / rate; GST_WARNING_OBJECT (self, "trying to re-initialize with period %i " "(%i frames, %i rate)", (int) device_period, n_frames, rate); hr = IAudioClient_Initialize (client, sharemode, stream_flags, device_period, device_period, format, NULL); } HR_FAILED_RET (hr, IAudioClient::Initialize, FALSE); if (sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) { /* We use the device period for the segment size and that needs to match * the buffer size exactly when we write into it */ hr = IAudioClient_GetBufferSize (client, &n_frames); HR_FAILED_RET (hr, IAudioClient::GetBufferSize, FALSE); *ret_devicep_frames = n_frames; } else { *ret_devicep_frames = (rate * device_period * 100) / GST_SECOND; } return TRUE; } gboolean gst_wasapi_util_initialize_audioclient3 (GstElement * self, GstAudioRingBufferSpec * spec, IAudioClient3 * client, WAVEFORMATEX * format, gboolean low_latency, gboolean loopback, guint * ret_devicep_frames) { HRESULT hr; gint stream_flags; guint devicep_frames; guint defaultp_frames, fundp_frames, minp_frames, maxp_frames; WAVEFORMATEX *tmpf; hr = IAudioClient3_GetSharedModeEnginePeriod (client, format, &defaultp_frames, &fundp_frames, &minp_frames, &maxp_frames); HR_FAILED_RET (hr, IAudioClient3::GetSharedModeEnginePeriod, FALSE); GST_INFO_OBJECT (self, "Using IAudioClient3, default period %i frames, " "fundamental period %i frames, minimum period %i frames, maximum period " "%i frames", defaultp_frames, fundp_frames, minp_frames, maxp_frames); if (low_latency) devicep_frames = minp_frames; else /* Just pick the max period, because lower values can cause glitches * https://bugzilla.gnome.org/show_bug.cgi?id=794497 */ devicep_frames = maxp_frames; stream_flags = AUDCLNT_STREAMFLAGS_EVENTCALLBACK; if (loopback) stream_flags |= AUDCLNT_STREAMFLAGS_LOOPBACK; hr = IAudioClient3_InitializeSharedAudioStream (client, stream_flags, devicep_frames, format, NULL); HR_FAILED_RET (hr, IAudioClient3::InitializeSharedAudioStream, FALSE); hr = IAudioClient3_GetCurrentSharedModeEnginePeriod (client, &tmpf, &devicep_frames); CoTaskMemFree (tmpf); HR_FAILED_RET (hr, IAudioClient3::GetCurrentSharedModeEnginePeriod, FALSE); *ret_devicep_frames = devicep_frames; return TRUE; }