gstreamer/sys/wasapi/gstwasapiutil.c
Nirbheek Chauhan 1450851095 wasapi: Rewrite most of the code to make it work
Both the source and the sink elements were broken in a number of ways:

* prepare() was assuming that the format was always S16LE 2ch 44.1KHz.
  We now probe the preferred format with GetMixFormat().
* Device initialization was done with the wrong buffer size
  (buffer_time is in microseconds, not nanoseconds).
* sink_write() and src_read() were just plain wrong and would never
  write or read anything useful.
* Some functions in prepare() were always returning FALSE which meant
  trying to use the elements would *always* fail.
* get_caps() and delay() were not implemented at all.

TODO: support for >2 channels
TODO: pro-audio low-latency
TODO: SPDIF and other encoded passthroughs

Three new properties are now implemented: role, mute, and device.

* 'role' designates the stream role of the initialized device, see:
   https://msdn.microsoft.com/en-us/library/windows/desktop/dd370842(v=vs.85).aspx
* 'device' is a system-wide GUIDesque string for a specific device.
* 'mute' is a sink property and simply mutes it.

On my Windows 8.1 system, the lowest latency that works is:

  wasapisrc buffer-time=20000
  wasapisink buffer-time=10000

aka, 20ms and 10ms respectively. These values are close to the lowest
possible with the IAudioClient interface. Further improvements require
porting to IAudioClient2 or IAudioClient3.

https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/low-latency-audio
2018-01-22 14:18:53 +05:30

412 lines
12 KiB
C

/*
* 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.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "gstwasapiutil.h"
#include <mmdeviceapi.h>
#ifdef __uuidof
const CLSID CLSID_MMDeviceEnumerator = __uuidof (MMDeviceEnumerator);
const IID IID_IMMDeviceEnumerator = __uuidof (IMMDeviceEnumerator);
const IID IID_IAudioClient = __uuidof (IAudioClient);
const IID IID_IAudioRenderClient = __uuidof (IAudioRenderClient);
const IID IID_IAudioCaptureClient = __uuidof (IAudioCaptureClient);
const IID IID_IAudioClock = __uuidof (IAudioClock);
#else
/* __uuidof is not implemented in our Cerbero's ancient MinGW toolchain 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_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32,
{0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2}
};
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}
};
#endif
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 ();
}
}
const gchar *
gst_wasapi_util_hresult_to_string (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_INVALIDARG:
s = "E_INVALIDARG";
break;
}
return s;
}
gboolean
gst_wasapi_util_get_device_client (GstElement * element,
gboolean capture, gint role, const wchar_t * device_name,
IAudioClient ** ret_client)
{
gboolean res = FALSE;
HRESULT hr;
IMMDeviceEnumerator *enumerator = NULL;
IMMDevice *device = NULL;
IAudioClient *client = NULL;
hr = CoCreateInstance (&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
&IID_IMMDeviceEnumerator, (void **) &enumerator);
if (hr != S_OK) {
GST_ERROR ("CoCreateInstance (MMDeviceEnumerator) failed: %s",
gst_wasapi_util_hresult_to_string (hr));
goto beach;
}
if (!device_name) {
hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint (enumerator,
capture ? eCapture : eRender, role, &device);
if (hr != S_OK) {
GST_ERROR ("IMMDeviceEnumerator::GetDefaultAudioEndpoint () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
goto beach;
}
} else {
hr = IMMDeviceEnumerator_GetDevice (enumerator, device_name, &device);
if (hr != S_OK) {
GST_ERROR ("IMMDeviceEnumerator::GetDevice (\"%S\") failed", device_name);
goto beach;
}
}
hr = IMMDevice_Activate (device, &IID_IAudioClient, CLSCTX_ALL, NULL,
(void **) &client);
if (hr != S_OK) {
GST_ERROR ("IMMDevice::Activate (IID_IAudioClient) failed: %s",
gst_wasapi_util_hresult_to_string (hr));
goto beach;
}
IUnknown_AddRef (client);
*ret_client = client;
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 * element, IAudioClient * client,
IAudioRenderClient ** ret_render_client)
{
gboolean res = FALSE;
HRESULT hr;
IAudioRenderClient *render_client = NULL;
hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
(void **) &render_client);
if (hr != S_OK) {
GST_ERROR ("IAudioClient::GetService (IID_IAudioRenderClient) failed: %s",
gst_wasapi_util_hresult_to_string (hr));
goto beach;
}
*ret_render_client = render_client;
res = TRUE;
beach:
return res;
}
gboolean
gst_wasapi_util_get_capture_client (GstElement * element, IAudioClient * client,
IAudioCaptureClient ** ret_capture_client)
{
gboolean res = FALSE;
HRESULT hr;
IAudioCaptureClient *capture_client = NULL;
hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
(void **) &capture_client);
if (hr != S_OK) {
GST_ERROR ("IAudioClient::GetService (IID_IAudioCaptureClient) failed: %s",
gst_wasapi_util_hresult_to_string (hr));
goto beach;
}
*ret_capture_client = capture_client;
res = TRUE;
beach:
return res;
}
gboolean
gst_wasapi_util_get_clock (GstElement * element, IAudioClient * client,
IAudioClock ** ret_clock)
{
gboolean res = FALSE;
HRESULT hr;
IAudioClock *clock = NULL;
hr = IAudioClient_GetService (client, &IID_IAudioClock, (void **) &clock);
if (hr != S_OK) {
GST_ERROR ("IAudioClient::GetService (IID_IAudioClock) failed: %s",
gst_wasapi_util_hresult_to_string (hr));
goto beach;
}
*ret_clock = clock;
res = TRUE;
beach:
return res;
}
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;
}
GstCaps *
gst_wasapi_util_waveformatex_to_caps (WAVEFORMATEXTENSIBLE * format,
GstCaps * template_caps)
{
int ii;
const gchar *afmt;
GstCaps *caps = gst_caps_copy (template_caps);
/* 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 NULL;
/* 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 NULL;
for (ii = 0; ii < gst_caps_get_size (caps); ii++) {
GstStructure *s = gst_caps_get_structure (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);
}
return caps;
}