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cubeb - re-enable and polish IAudioClient3 to achieve lower latencies

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Filoppi 2023-01-20 20:55:20 +02:00 коммит произвёл Paul Adenot
Родитель 1ba9237364
Коммит 0c0c88952d
1 изменённых файлов: 185 добавлений и 63 удалений
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248
src/cubeb_wasapi.cpp
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@ -37,6 +37,31 @@
#include "cubeb_tracing.h"
#include "cubeb_utils.h"
// Some people have reported glitches with IAudioClient3 capture streams:
// http://blog.nirbheek.in/2018/03/low-latency-audio-on-windows-with.html
// https://bugzilla.mozilla.org/show_bug.cgi?id=1590902
#define ALLOW_AUDIO_CLIENT_3_FOR_INPUT 0
// IAudioClient3::GetSharedModeEnginePeriod() seem to return min latencies
// bigger than IAudioClient::GetDevicePeriod(), which is confusing (10ms vs
// 3ms), though the default latency is usually the same and we should use the
// IAudioClient3 function anyway, as it's more correct
#define USE_AUDIO_CLIENT_3_MIN_PERIOD 1
// If this is true, we allow IAudioClient3 the creation of sessions with a
// latency above the default one (usually 10ms).
// Whether we should default this to true or false depend on many things:
// -Does creating a shared IAudioClient3 session (not locked to a format)
// actually forces all the IAudioClient(1) sessions to have the same latency?
// I could find no proof of that.
// -Does creating a shared IAudioClient3 session with a latency >= the default
// one actually improve the latency (as in how late the audio is) at all?
// -Maybe we could expose this as cubeb stream pref
// (e.g. take priority over other apps)?
#define ALLOW_AUDIO_CLIENT_3_LATENCY_OVER_DEFAULT 1
// If this is true and the user specified a target latency >= the IAudioClient3
// max one, then we reject it and fall back to IAudioClient(1). There wouldn't
// be much point in having a low latency if that's not what the user wants.
#define REJECT_AUDIO_CLIENT_3_LATENCY_OVER_MAX 0
// Windows 10 exposes the IAudioClient3 interface to create low-latency streams.
// Copy the interface definition from audioclient.h here to make the code
// simpler and so that we can still access IAudioClient3 via COM if cubeb was
@ -1893,6 +1918,44 @@ wasapi_get_min_latency(cubeb * ctx, cubeb_stream_params params,
return CUBEB_ERROR;
}
#if USE_AUDIO_CLIENT_3_MIN_PERIOD
// This is unreliable as we can't know the actual mixer format cubeb will
// ask for later on (nor we can branch on ALLOW_AUDIO_CLIENT_3_FOR_INPUT),
// and the min latency can change based on that.
com_ptr<IAudioClient3> client3;
hr = device->Activate(__uuidof(IAudioClient3), CLSCTX_INPROC_SERVER, NULL,
client3.receive_vpp());
if (SUCCEEDED(hr)) {
WAVEFORMATEX * mix_format = nullptr;
hr = client3->GetMixFormat(&mix_format);
if (SUCCEEDED(hr)) {
uint32_t default_period = 0, fundamental_period = 0, min_period = 0,
max_period = 0;
hr = client3->GetSharedModeEnginePeriod(mix_format, &default_period,
&fundamental_period, &min_period,
&max_period);
auto sample_rate = mix_format->nSamplesPerSec;
CoTaskMemFree(mix_format);
if (SUCCEEDED(hr)) {
// Print values in the same format as IAudioDevice::GetDevicePeriod()
REFERENCE_TIME min_period_rt(frames_to_hns(sample_rate, min_period));
REFERENCE_TIME default_period_rt(
frames_to_hns(sample_rate, default_period));
LOG("default device period: %I64d, minimum device period: %I64d",
default_period_rt, min_period_rt);
*latency_frames = min_period;
LOG("Minimum latency in frames: %u", *latency_frames);
return CUBEB_OK;
}
}
}
#endif
com_ptr<IAudioClient> client;
hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
client.receive_vpp());
@ -1912,18 +1975,8 @@ wasapi_get_min_latency(cubeb * ctx, cubeb_stream_params params,
LOG("default device period: %I64d, minimum device period: %I64d",
default_period, minimum_period);
/* If we're on Windows 10, we can use IAudioClient3 to get minimal latency.
Otherwise, according to the docs, the best latency we can achieve is by
synchronizing the stream and the engine.
http://msdn.microsoft.com/en-us/library/windows/desktop/dd370871%28v=vs.85%29.aspx
*/
// #ifdef _WIN32_WINNT_WIN10
#if 0
*latency_frames = hns_to_frames(params.rate, minimum_period);
#else
// The minimum_period is only relevant in exclusive streams.
*latency_frames = hns_to_frames(params.rate, default_period);
#endif
LOG("Minimum latency in frames: %u", *latency_frames);
@ -2013,7 +2066,10 @@ handle_channel_layout(cubeb_stream * stm, EDataFlow direction,
if (hr == S_FALSE) {
/* Channel layout not supported, but WASAPI gives us a suggestion. Use it,
and handle the eventual upmix/downmix ourselves. Ignore the subformat of
the suggestion, since it seems to always be IEEE_FLOAT. */
the suggestion, since it seems to always be IEEE_FLOAT.
This fallback doesn't update the bit depth, so if a device
only supported bit depths cubeb doesn't support, so IAudioClient3
streams might fail */
LOG("Using WASAPI suggested format: channels: %d", closest->nChannels);
XASSERT(closest->wFormatTag == WAVE_FORMAT_EXTENSIBLE);
WAVEFORMATEXTENSIBLE * closest_pcm =
@ -2057,12 +2113,12 @@ initialize_iaudioclient2(com_ptr<IAudioClient> & audio_client)
return CUBEB_OK;
}
#if 0
bool
initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
cubeb_stream * stm,
const com_heap_ptr<WAVEFORMATEX> & mix_format,
DWORD flags, EDataFlow direction)
DWORD flags, EDataFlow direction,
REFERENCE_TIME latency_hns)
{
com_ptr<IAudioClient3> audio_client3;
audio_client->QueryInterface<IAudioClient3>(audio_client3.receive());
@ -2078,24 +2134,22 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
return false;
}
// Some people have reported glitches with capture streams:
// http://blog.nirbheek.in/2018/03/low-latency-audio-on-windows-with.html
if (direction == eCapture) {
LOG("Audio stream is capture, not using IAudioClient3");
return false;
}
// Possibly initialize a shared-mode stream using IAudioClient3. Initializing
// a stream this way lets you request lower latencies, but also locks the
// global WASAPI engine at that latency.
// - If we request a shared-mode stream, streams created with IAudioClient
// will
// have their latency adjusted to match. When the shared-mode stream is
// closed, they'll go back to normal.
// - If there's already a shared-mode stream running, then we cannot request
// the engine change to a different latency - we have to match it.
// - It's antisocial to lock the WASAPI engine at its default latency. If we
// would do this, then stop and use IAudioClient instead.
// might have their latency adjusted to match. When the shared-mode stream
// is closed, they'll go back to normal.
// - If there's already a shared-mode stream running, if it created with the
// AUDCLNT_STREAMOPTIONS_MATCH_FORMAT option, the audio engine would be
// locked to that format, so we have to match it (a custom one would fail).
// - We don't lock the WASAPI engine to a format, as it's antisocial towards
// other apps, especially if we locked to a latency >= than its default.
// - If the user requested latency is >= the default one, we might still
// accept it (without locking the format) depending on
// ALLOW_AUDIO_CLIENT_3_LATENCY_OVER_DEFAULT, as we might want to prioritize
// to lower our latency over other apps
// (there might still be latency advantages compared to IAudioDevice(1)).
HRESULT hr;
uint32_t default_period = 0, fundamental_period = 0, min_period = 0,
@ -2107,28 +2161,59 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
LOG("Could not get shared mode engine period: error: %lx", hr);
return false;
}
uint32_t requested_latency = stm->latency;
uint32_t requested_latency =
hns_to_frames(mix_format->nSamplesPerSec, latency_hns);
#if !ALLOW_AUDIO_CLIENT_3_LATENCY_OVER_DEFAULT
if (requested_latency >= default_period) {
LOG("Requested latency %i greater than default latency %i, not using "
"IAudioClient3",
LOG("Requested latency %i equal or greater than default latency %i,"
" not using IAudioClient3",
requested_latency, default_period);
return false;
}
#elif REJECT_AUDIO_CLIENT_3_LATENCY_OVER_MAX
if (requested_latency > max_period) {
// Fallback to IAudioClient(1) as it's more accepting of large latencies
LOG("Requested latency %i greater than max latency %i,"
" not using IAudioClient3",
requested_latency, max_period);
return false;
}
#endif
LOG("Got shared mode engine period: default=%i fundamental=%i min=%i max=%i",
default_period, fundamental_period, min_period, max_period);
// Snap requested latency to a valid value
uint32_t old_requested_latency = requested_latency;
// The period is required to be a multiple of the fundamental period
// (and >= min and <= max, which should still be true)
requested_latency -= requested_latency % fundamental_period;
if (requested_latency < min_period) {
requested_latency = min_period;
}
requested_latency -= (requested_latency - min_period) % fundamental_period;
// Likely unnecessary, but won't hurt
if (requested_latency > max_period) {
requested_latency = max_period;
}
if (requested_latency != old_requested_latency) {
LOG("Requested latency %i was adjusted to %i", old_requested_latency,
requested_latency);
}
hr = audio_client3->InitializeSharedAudioStream(flags, requested_latency,
DWORD new_flags = flags;
// Always add these flags to IAudioClient3, they might help
// if the stream doesn't have the same format as the audio engine.
new_flags |= AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM;
new_flags |= AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
hr = audio_client3->InitializeSharedAudioStream(new_flags, requested_latency,
mix_format.get(), NULL);
// This error should be returned first even if
// the period was locked (AUDCLNT_E_ENGINE_PERIODICITY_LOCKED)
if (hr == AUDCLNT_E_INVALID_STREAM_FLAG) {
LOG("Got AUDCLNT_E_INVALID_STREAM_FLAG, removing some flags");
hr = audio_client3->InitializeSharedAudioStream(flags, requested_latency,
mix_format.get(), NULL);
}
if (SUCCEEDED(hr)) {
return true;
} else if (hr == AUDCLNT_E_ENGINE_PERIODICITY_LOCKED) {
@ -2140,23 +2225,38 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
}
uint32_t current_period = 0;
WAVEFORMATEX * current_format = nullptr;
WAVEFORMATEX * current_format_ptr = nullptr;
// We have to pass a valid WAVEFORMATEX** and not nullptr, otherwise
// GetCurrentSharedModeEnginePeriod will return E_POINTER
hr = audio_client3->GetCurrentSharedModeEnginePeriod(&current_format,
hr = audio_client3->GetCurrentSharedModeEnginePeriod(&current_format_ptr,
&current_period);
CoTaskMemFree(current_format);
if (FAILED(hr)) {
LOG("Could not get current shared mode engine period: error: %lx", hr);
return false;
}
if (current_period >= default_period) {
LOG("Current shared mode engine period %i too high, not using IAudioClient",
current_period);
com_heap_ptr<WAVEFORMATEX> current_format(current_format_ptr);
if (current_format->nSamplesPerSec != mix_format->nSamplesPerSec) {
// Unless some other external app locked the shared mode engine period
// within our audio initialization, this is unlikely to happen, though we
// can't respect the user selected latency, so we fallback on IAudioClient
LOG("IAudioClient3::GetCurrentSharedModeEnginePeriod() returned a "
"different mixer format (nSamplesPerSec) from "
"IAudioClient::GetMixFormat(); not using IAudioClient3");
return false;
}
#if REJECT_AUDIO_CLIENT_3_LATENCY_OVER_MAX
// Reject IAudioClient3 if we can't respect the user target latency.
// We don't need to check against default_latency anymore,
// as the current_period is already the best one we could get.
if (old_requested_latency > current_period) {
LOG("Requested latency %i greater than currently locked shared mode "
"latency %i, not using IAudioClient3",
old_requested_latency, current_period);
return false;
}
#endif
hr = audio_client3->InitializeSharedAudioStream(flags, current_period,
mix_format.get(), NULL);
if (SUCCEEDED(hr)) {
@ -2168,7 +2268,6 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
LOG("Could not initialize shared stream with IAudioClient3: error: %lx", hr);
return false;
}
#endif
#define DIRECTION_NAME (direction == eCapture ? "capture" : "render")
@ -2192,6 +2291,12 @@ setup_wasapi_stream_one_side(cubeb_stream * stm,
return CUBEB_ERROR;
}
#if ALLOW_AUDIO_CLIENT_3_FOR_INPUT
constexpr bool allow_audio_client_3 = true;
#else
const bool allow_audio_client_3 = direction == eRender;
#endif
stm->stream_reset_lock.assert_current_thread_owns();
// If user doesn't specify a particular device, we can choose another one when
// the given devid is unavailable.
@ -2228,17 +2333,14 @@ setup_wasapi_stream_one_side(cubeb_stream * stm,
/* Get a client. We will get all other interfaces we need from
* this pointer. */
#if 0 // See https://bugzilla.mozilla.org/show_bug.cgi?id=1590902
hr = device->Activate(__uuidof(IAudioClient3),
CLSCTX_INPROC_SERVER,
NULL, audio_client.receive_vpp());
if (hr == E_NOINTERFACE) {
#endif
hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
audio_client.receive_vpp());
#if 0
if (allow_audio_client_3) {
hr = device->Activate(__uuidof(IAudioClient3), CLSCTX_INPROC_SERVER, NULL,
audio_client.receive_vpp());
}
if (!allow_audio_client_3 || hr == E_NOINTERFACE) {
hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
audio_client.receive_vpp());
}
#endif
if (FAILED(hr)) {
LOG("Could not activate the device to get an audio"
@ -2367,16 +2469,15 @@ setup_wasapi_stream_one_side(cubeb_stream * stm,
}
}
#if 0 // See https://bugzilla.mozilla.org/show_bug.cgi?id=1590902
if (initialize_iaudioclient3(audio_client, stm, mix_format, flags, direction)) {
if (allow_audio_client_3 &&
initialize_iaudioclient3(audio_client, stm, mix_format, flags, direction,
latency_hns)) {
LOG("Initialized with IAudioClient3");
} else {
#endif
hr = audio_client->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, latency_hns, 0,
mix_format.get(), NULL);
#if 0
hr = audio_client->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, latency_hns,
0, mix_format.get(), NULL);
}
#endif
if (FAILED(hr)) {
LOG("Unable to initialize audio client for %s: %lx.", DIRECTION_NAME, hr);
return CUBEB_ERROR;
@ -3334,6 +3435,7 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
CUBEB_DEVICE_FMT_S16NE);
ret.default_format = CUBEB_DEVICE_FMT_F32NE;
prop_variant fmtvar;
WAVEFORMATEX * wfx = NULL;
hr = propstore->GetValue(PKEY_AudioEngine_DeviceFormat, &fmtvar);
if (SUCCEEDED(hr) && fmtvar.vt == VT_BLOB) {
if (fmtvar.blob.cbSize == sizeof(PCMWAVEFORMAT)) {
@ -3343,8 +3445,7 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
ret.max_rate = ret.min_rate = ret.default_rate = pcm->wf.nSamplesPerSec;
ret.max_channels = pcm->wf.nChannels;
} else if (fmtvar.blob.cbSize >= sizeof(WAVEFORMATEX)) {
WAVEFORMATEX * wfx =
reinterpret_cast<WAVEFORMATEX *>(fmtvar.blob.pBlobData);
wfx = reinterpret_cast<WAVEFORMATEX *>(fmtvar.blob.pBlobData);
if (fmtvar.blob.cbSize >= sizeof(WAVEFORMATEX) + wfx->cbSize ||
wfx->wFormatTag == WAVE_FORMAT_PCM) {
@ -3354,9 +3455,30 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
}
}
if (SUCCEEDED(dev->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER,
NULL, client.receive_vpp())) &&
SUCCEEDED(client->GetDevicePeriod(&def_period, &min_period))) {
#if USE_AUDIO_CLIENT_3_MIN_PERIOD
// Here we assume an IAudioClient3 stream will successfully
// be initialized later (it might fail)
#if ALLOW_AUDIO_CLIENT_3_FOR_INPUT
constexpr bool allow_audio_client_3 = true;
#else
const bool allow_audio_client_3 = flow == eRender;
#endif
com_ptr<IAudioClient3> client3;
uint32_t def, fun, min, max;
if (allow_audio_client_3 && wfx &&
SUCCEEDED(dev->Activate(__uuidof(IAudioClient3), CLSCTX_INPROC_SERVER,
NULL, client3.receive_vpp())) &&
SUCCEEDED(
client3->GetSharedModeEnginePeriod(wfx, &def, &fun, &min, &max))) {
ret.latency_lo = min;
// This latency might actually be used as "default" and not "max" later on,
// so we return the default (we never really want to use the max anyway)
ret.latency_hi = def;
} else
#endif
if (SUCCEEDED(dev->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER,
NULL, client.receive_vpp())) &&
SUCCEEDED(client->GetDevicePeriod(&def_period, &min_period))) {
ret.latency_lo = hns_to_frames(ret.default_rate, min_period);
ret.latency_hi = hns_to_frames(ret.default_rate, def_period);
} else {