diff --git a/src/cubeb_wasapi.cpp b/src/cubeb_wasapi.cpp index 7767b3a2..68deb81f 100644 --- a/src/cubeb_wasapi.cpp +++ b/src/cubeb_wasapi.cpp @@ -33,6 +33,31 @@ #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 @@ -1867,6 +1892,44 @@ 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 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 client; hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL, client.receive_vpp()); @@ -1886,18 +1949,8 @@ 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); @@ -1987,7 +2040,10 @@ 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 = @@ -2031,12 +2087,12 @@ initialize_iaudioclient2(com_ptr & audio_client) return CUBEB_OK; } -#if 0 bool initialize_iaudioclient3(com_ptr & audio_client, cubeb_stream * stm, const com_heap_ptr & mix_format, - DWORD flags, EDataFlow direction) + DWORD flags, EDataFlow direction, + REFERENCE_TIME latency_hns) { com_ptr audio_client3; audio_client->QueryInterface(audio_client3.receive()); @@ -2052,24 +2108,22 @@ initialize_iaudioclient3(com_ptr & 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, @@ -2081,28 +2135,59 @@ initialize_iaudioclient3(com_ptr & 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) { @@ -2114,22 +2199,37 @@ initialize_iaudioclient3(com_ptr & 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(¤t_format, + hr = audio_client3->GetCurrentSharedModeEnginePeriod(¤t_format_ptr, ¤t_period); - CoTaskMemFree(current_format); if (FAILED(hr)) { LOG("Could not get current shared mode engine period: error: %lx", hr); return false; } + com_heap_ptr 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 (current_period >= default_period) { - LOG("Current shared mode engine period %i too high, not using IAudioClient", - current_period); +#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); @@ -2142,7 +2242,6 @@ initialize_iaudioclient3(com_ptr & audio_client, LOG("Could not initialize shared stream with IAudioClient3: error: %lx", hr); return false; } -#endif #define DIRECTION_NAME (direction == eCapture ? "capture" : "render") @@ -2166,6 +2265,12 @@ 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. @@ -2202,17 +2307,14 @@ 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" @@ -2341,16 +2443,15 @@ 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; @@ -3310,6 +3411,7 @@ 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)) { @@ -3319,8 +3421,7 @@ 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(fmtvar.blob.pBlobData); + wfx = reinterpret_cast(fmtvar.blob.pBlobData); if (fmtvar.blob.cbSize >= sizeof(WAVEFORMATEX) + wfx->cbSize || wfx->wFormatTag == WAVE_FORMAT_PCM) { @@ -3330,9 +3431,30 @@ 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 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 {