3rdparty/cubeb: Sync to 19fcbef

Backport of 872cee908c

And apply PR #740 (Re-enable and polish IAudioClient3 to achieve lower
latencies).

`*latency_frames = min_period;` in wasapi_get_min_latency was changed to
`*latency_frames = hns_to_frames(params.rate, min_period_rt);`, as
otherwise it reports in mixer frames, not stream frames.

3rdparty/cubeb/CMakeLists.txt

@@ -37,15 +37,15 @@ target_include_directories(cubeb
   PUBLIC $<BUILD_INTERFACE:${CMAKE_CURRENT_SOURCE_DIR}/include> $<INSTALL_INTERFACE:include>
 )
 
-add_library(speex OBJECT subprojects/speex/resample.c)
+  add_library(speex OBJECT subprojects/speex/resample.c)
-set_target_properties(speex PROPERTIES POSITION_INDEPENDENT_CODE TRUE)
+  set_target_properties(speex PROPERTIES POSITION_INDEPENDENT_CODE TRUE)
-target_include_directories(speex INTERFACE subprojects)
+  target_include_directories(speex INTERFACE subprojects)
-target_compile_definitions(speex PUBLIC
+  target_compile_definitions(speex PUBLIC
-  OUTSIDE_SPEEX
+    OUTSIDE_SPEEX
-  FLOATING_POINT
+    FLOATING_POINT
-  EXPORT=
+    EXPORT=
-  RANDOM_PREFIX=speex
+    RANDOM_PREFIX=speex
-)
+  )
 
 # $<BUILD_INTERFACE:> required because of https://gitlab.kitware.com/cmake/cmake/-/issues/15415
 target_link_libraries(cubeb PRIVATE $<BUILD_INTERFACE:speex>)
@@ -58,17 +58,20 @@ find_package(Threads)
 target_link_libraries(cubeb PRIVATE Threads::Threads)
 
 if(LAZY_LOAD_LIBS)
-  if(NOT APPLE AND NOT WIN32)
+  if(NOT APPLE AND NOT WIN32) # Skip checks on MacOS because it takes ages in XCode.
-    # Skip checks on MacOS because it takes ages in XCode.
+  check_include_files(pulse/pulseaudio.h USE_PULSE)
-    check_include_files(pulse/pulseaudio.h USE_PULSE)
+  check_include_files(alsa/asoundlib.h   USE_ALSA)
-    check_include_files(alsa/asoundlib.h   USE_ALSA)
+  check_include_files(jack/jack.h        USE_JACK)
-    check_include_files(jack/jack.h        USE_JACK)
+  check_include_files(sndio.h            USE_SNDIO)
-    check_include_files(sndio.h            USE_SNDIO)
 
-    if(USE_PULSE OR USE_ALSA OR USE_JACK OR USE_SNDIO)
+  if(USE_PULSE OR USE_ALSA OR USE_JACK OR USE_SNDIO OR USE_AAUDIO)
-      target_link_libraries(cubeb PRIVATE ${CMAKE_DL_LIBS})
+    target_link_libraries(cubeb PRIVATE ${CMAKE_DL_LIBS})
+
+    if(ANDROID)
+      target_compile_definitions(cubeb PRIVATE __ANDROID_UNAVAILABLE_SYMBOLS_ARE_WEAK__)
     endif()
   endif()
+  endif()
 
 elseif(NOT APPLE AND NOT WIN32)
 
@@ -123,56 +126,56 @@ if(USE_SNDIO)
 endif()
 
 if(APPLE)
-  check_include_files(AudioUnit/AudioUnit.h USE_AUDIOUNIT)
+check_include_files(AudioUnit/AudioUnit.h USE_AUDIOUNIT)
-  if(USE_AUDIOUNIT)
+if(USE_AUDIOUNIT)
-    target_sources(cubeb PRIVATE
+  target_sources(cubeb PRIVATE
-      src/cubeb_audiounit.cpp
+    src/cubeb_audiounit.cpp
-      src/cubeb_osx_run_loop.cpp)
+    src/cubeb_osx_run_loop.cpp)
-    target_compile_definitions(cubeb PRIVATE USE_AUDIOUNIT)
+  target_compile_definitions(cubeb PRIVATE USE_AUDIOUNIT)
-    target_link_libraries(cubeb PRIVATE "-framework AudioUnit" "-framework CoreAudio" "-framework CoreServices")
+  target_link_libraries(cubeb PRIVATE "-framework AudioUnit" "-framework CoreAudio" "-framework CoreServices")
-  endif()
+endif()
 endif()
 
 if(WIN32)
-  check_include_files(audioclient.h USE_WASAPI)
+check_include_files(audioclient.h USE_WASAPI)
-  if(USE_WASAPI)
+if(USE_WASAPI)
-    target_sources(cubeb PRIVATE
+  target_sources(cubeb PRIVATE
-      src/cubeb_wasapi.cpp)
+    src/cubeb_wasapi.cpp)
-    target_compile_definitions(cubeb PRIVATE USE_WASAPI)
+  target_compile_definitions(cubeb PRIVATE USE_WASAPI)
-    target_link_libraries(cubeb PRIVATE avrt ole32 ksuser)
+  target_link_libraries(cubeb PRIVATE ole32 ksuser)
-  endif()
+endif()
 
-  check_include_files("windows.h;mmsystem.h" USE_WINMM)
+check_include_files("windows.h;mmsystem.h" USE_WINMM)
-  if(USE_WINMM)
+if(USE_WINMM)
-    target_sources(cubeb PRIVATE
+  target_sources(cubeb PRIVATE
-      src/cubeb_winmm.c)
+    src/cubeb_winmm.c)
-    target_compile_definitions(cubeb PRIVATE USE_WINMM)
+  target_compile_definitions(cubeb PRIVATE USE_WINMM)
-    target_link_libraries(cubeb PRIVATE winmm)
+  target_link_libraries(cubeb PRIVATE winmm)
-  endif()
+endif()
 endif()
 
 if(NOT WIN32 AND NOT APPLE)
-  check_include_files(sys/soundcard.h HAVE_SYS_SOUNDCARD_H)
+check_include_files(sys/soundcard.h HAVE_SYS_SOUNDCARD_H)
-  if(HAVE_SYS_SOUNDCARD_H)
+if(HAVE_SYS_SOUNDCARD_H)
-    try_compile(USE_OSS "${PROJECT_BINARY_DIR}/compile_tests"
+  try_compile(USE_OSS "${PROJECT_BINARY_DIR}/compile_tests"
-      ${PROJECT_SOURCE_DIR}/cmake/compile_tests/oss_is_v4.c)
+    ${PROJECT_SOURCE_DIR}/cmake/compile_tests/oss_is_v4.c)
-    if(USE_OSS)
+  if(USE_OSS)
-      # strlcpy is not available on BSD systems that use glibc,
+    # strlcpy is not available on BSD systems that use glibc,
-      # like Debian kfreebsd, so try using libbsd if available
+    # like Debian kfreebsd, so try using libbsd if available
-      include(CheckSymbolExists)
+    include(CheckSymbolExists)
-      check_symbol_exists(strlcpy string.h HAVE_STRLCPY)
+    check_symbol_exists(strlcpy string.h HAVE_STRLCPY)
-      if(NOT HAVE_STRLCPY)
+    if(NOT HAVE_STRLCPY)
-        pkg_check_modules(libbsd-overlay IMPORTED_TARGET libbsd-overlay)
+      pkg_check_modules(libbsd-overlay IMPORTED_TARGET libbsd-overlay)
-        if(libbsd-overlay_FOUND)
+      if(libbsd-overlay_FOUND)
-          target_link_libraries(cubeb PRIVATE PkgConfig::libbsd-overlay)
+        target_link_libraries(cubeb PRIVATE PkgConfig::libbsd-overlay)
-          set(HAVE_STRLCPY true)
+        set(HAVE_STRLCPY true)
-        endif()
-      endif()
-      if (HAVE_STRLCPY)
-        target_sources(cubeb PRIVATE
-          src/cubeb_oss.c)
-        target_compile_definitions(cubeb PRIVATE USE_OSS)
       endif()
     endif()
+    if (HAVE_STRLCPY)
+      target_sources(cubeb PRIVATE
+        src/cubeb_oss.c)
+      target_compile_definitions(cubeb PRIVATE USE_OSS)
+    endif()
   endif()
 endif()
+endif()

3rdparty/cubeb/include/cubeb/cubeb.h

@@ -258,11 +258,23 @@ typedef enum {
                                                    the jack backend. */
 } cubeb_stream_prefs;
 
+/**
+ * Input stream audio processing parameters. Only applicable with
+ * CUBEB_STREAM_PREF_VOICE.
+ */
+typedef enum {
+  CUBEB_INPUT_PROCESSING_PARAM_NONE = 0x00,
+  CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION = 0x01,
+  CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION = 0x02,
+  CUBEB_INPUT_PROCESSING_PARAM_AUTOMATIC_GAIN_CONTROL = 0x04,
+  CUBEB_INPUT_PROCESSING_PARAM_VOICE_ISOLATION = 0x08,
+} cubeb_input_processing_params;
+
 /** Stream format initialization parameters. */
 typedef struct {
   cubeb_sample_format format; /**< Requested sample format.  One of
                                    #cubeb_sample_format. */
-  uint32_t rate; /**< Requested sample rate.  Valid range is [1000, 192000]. */
+  uint32_t rate; /**< Requested sample rate.  Valid range is [1000, 384000]. */
   uint32_t channels; /**< Requested channel count.  Valid range is [1, 8]. */
   cubeb_channel_layout
       layout; /**< Requested channel layout. This must be consistent with the
@@ -519,6 +531,18 @@ cubeb_get_min_latency(cubeb * context, cubeb_stream_params * params,
 CUBEB_EXPORT int
 cubeb_get_preferred_sample_rate(cubeb * context, uint32_t * rate);
 
+/** Get the supported input processing features for this backend. See
+    cubeb_stream_set_input_processing for how to set them for a particular input
+    stream.
+    @param context A pointer to the cubeb context.
+    @param params Out parameter for the input processing params supported by
+                  this backend.
+    @retval CUBEB_OK
+    @retval CUBEB_ERROR_NOT_SUPPORTED */
+CUBEB_EXPORT int
+cubeb_get_supported_input_processing_params(
+    cubeb * context, cubeb_input_processing_params * params);
+
 /** Destroy an application context. This must be called after all stream have
  *  been destroyed.
     @param context A pointer to the cubeb context.*/
@@ -646,6 +670,30 @@ CUBEB_EXPORT int
 cubeb_stream_get_current_device(cubeb_stream * stm,
                                 cubeb_device ** const device);
 
+/** Set input mute state for this stream. Some platforms notify the user when an
+    application is accessing audio input. When all inputs are muted they can
+    prove to the user that the application is not actively capturing any input.
+    @param stream the stream for which to set input mute state
+    @param muted whether the input should mute or not
+    @retval CUBEB_OK
+    @retval CUBEB_ERROR_INVALID_PARAMETER if this stream does not have an input
+            device
+    @retval CUBEB_ERROR_NOT_SUPPORTED */
+CUBEB_EXPORT int
+cubeb_stream_set_input_mute(cubeb_stream * stream, int mute);
+
+/** Set what input processing features to enable for this stream.
+    @param stream the stream for which to set input processing features.
+    @param params what input processing features to use
+    @retval CUBEB_OK
+    @retval CUBEB_ERROR if params could not be applied
+    @retval CUBEB_ERROR_INVALID_PARAMETER if a given param is not supported by
+            this backend, or if this stream does not have an input device
+    @retval CUBEB_ERROR_NOT_SUPPORTED */
+CUBEB_EXPORT int
+cubeb_stream_set_input_processing_params(cubeb_stream * stream,
+                                         cubeb_input_processing_params params);
+
 /** Destroy a cubeb_device structure.
     @param stream the stream passed in cubeb_stream_get_current_device
     @param devices the devices to destroy

3rdparty/cubeb/src/cubeb-internal.h

@@ -40,6 +40,8 @@ struct cubeb_ops {
   int (*get_min_latency)(cubeb * context, cubeb_stream_params params,
                          uint32_t * latency_ms);
   int (*get_preferred_sample_rate)(cubeb * context, uint32_t * rate);
+  int (*get_supported_input_processing_params)(
+      cubeb * context, cubeb_input_processing_params * params);
   int (*enumerate_devices)(cubeb * context, cubeb_device_type type,
                            cubeb_device_collection * collection);
   int (*device_collection_destroy)(cubeb * context,
@@ -62,6 +64,9 @@ struct cubeb_ops {
   int (*stream_set_name)(cubeb_stream * stream, char const * stream_name);
   int (*stream_get_current_device)(cubeb_stream * stream,
                                    cubeb_device ** const device);
+  int (*stream_set_input_mute)(cubeb_stream * stream, int mute);
+  int (*stream_set_input_processing_params)(
+      cubeb_stream * stream, cubeb_input_processing_params params);
   int (*stream_device_destroy)(cubeb_stream * stream, cubeb_device * device);
   int (*stream_register_device_changed_callback)(
       cubeb_stream * stream,

3rdparty/cubeb/src/cubeb.c

@@ -67,7 +67,7 @@ validate_stream_params(cubeb_stream_params * input_stream_params,
   XASSERT(input_stream_params || output_stream_params);
   if (output_stream_params) {
     if (output_stream_params->rate < 1000 ||
-        output_stream_params->rate > 192000 ||
+        output_stream_params->rate > 768000 ||
         output_stream_params->channels < 1 ||
         output_stream_params->channels > UINT8_MAX) {
       return CUBEB_ERROR_INVALID_FORMAT;
@@ -75,7 +75,7 @@ validate_stream_params(cubeb_stream_params * input_stream_params,
   }
   if (input_stream_params) {
     if (input_stream_params->rate < 1000 ||
-        input_stream_params->rate > 192000 ||
+        input_stream_params->rate > 768000 ||
         input_stream_params->channels < 1 ||
         input_stream_params->channels > UINT8_MAX) {
       return CUBEB_ERROR_INVALID_FORMAT;
@@ -239,9 +239,6 @@ cubeb_get_backend_names()
 #if defined(USE_SNDIO)
     "sndio",
 #endif
-#if defined(USE_SUN)
-    "sun",
-#endif
 #if defined(USE_OSS)
     "oss",
 #endif
@@ -304,6 +301,21 @@ cubeb_get_preferred_sample_rate(cubeb * context, uint32_t * rate)
   return context->ops->get_preferred_sample_rate(context, rate);
 }
 
+int
+cubeb_get_supported_input_processing_params(
+    cubeb * context, cubeb_input_processing_params * params)
+{
+  if (!context || !params) {
+    return CUBEB_ERROR_INVALID_PARAMETER;
+  }
+
+  if (!context->ops->get_supported_input_processing_params) {
+    return CUBEB_ERROR_NOT_SUPPORTED;
+  }
+
+  return context->ops->get_supported_input_processing_params(context, params);
+}
+
 void
 cubeb_destroy(cubeb * context)
 {
@@ -463,6 +475,36 @@ cubeb_stream_get_current_device(cubeb_stream * stream,
   return stream->context->ops->stream_get_current_device(stream, device);
 }
 
+int
+cubeb_stream_set_input_mute(cubeb_stream * stream, int mute)
+{
+  if (!stream) {
+    return CUBEB_ERROR_INVALID_PARAMETER;
+  }
+
+  if (!stream->context->ops->stream_set_input_mute) {
+    return CUBEB_ERROR_NOT_SUPPORTED;
+  }
+
+  return stream->context->ops->stream_set_input_mute(stream, mute);
+}
+
+int
+cubeb_stream_set_input_processing_params(cubeb_stream * stream,
+                                         cubeb_input_processing_params params)
+{
+  if (!stream) {
+    return CUBEB_ERROR_INVALID_PARAMETER;
+  }
+
+  if (!stream->context->ops->stream_set_input_processing_params) {
+    return CUBEB_ERROR_NOT_SUPPORTED;
+  }
+
+  return stream->context->ops->stream_set_input_processing_params(stream,
+                                                                  params);
+}
+
 int
 cubeb_stream_device_destroy(cubeb_stream * stream, cubeb_device * device)
 {

3rdparty/cubeb/src/cubeb_alsa.c

@@ -1472,6 +1472,7 @@ static struct cubeb_ops const alsa_ops = {
     .get_max_channel_count = alsa_get_max_channel_count,
     .get_min_latency = alsa_get_min_latency,
     .get_preferred_sample_rate = alsa_get_preferred_sample_rate,
+    .get_supported_input_processing_params = NULL,
     .enumerate_devices = alsa_enumerate_devices,
     .device_collection_destroy = alsa_device_collection_destroy,
     .destroy = alsa_destroy,
@@ -1485,6 +1486,8 @@ static struct cubeb_ops const alsa_ops = {
     .stream_set_volume = alsa_stream_set_volume,
     .stream_set_name = NULL,
     .stream_get_current_device = NULL,
+    .stream_set_input_mute = NULL,
+    .stream_set_input_processing_params = NULL,
     .stream_device_destroy = NULL,
     .stream_register_device_changed_callback = NULL,
     .register_device_collection_changed = NULL};

3rdparty/cubeb/src/cubeb_audiounit.cpp

@@ -377,6 +377,15 @@ cubeb_channel_to_channel_label(cubeb_channel channel)
   }
 }
 
+bool
+is_common_sample_rate(Float64 sample_rate)
+{
+  /* Some commonly used sample rates and their multiples and divisors. */
+  return sample_rate == 8000 || sample_rate == 16000 || sample_rate == 22050 ||
+         sample_rate == 32000 || sample_rate == 44100 || sample_rate == 48000 ||
+         sample_rate == 88200 || sample_rate == 96000;
+}
+
 #if TARGET_OS_IPHONE
 typedef UInt32 AudioDeviceID;
 typedef UInt32 AudioObjectID;
@@ -2502,6 +2511,12 @@ audiounit_configure_output(cubeb_stream * stm)
     return CUBEB_ERROR;
   }
   stm->output_hw_rate = output_hw_desc.mSampleRate;
+  if (!is_common_sample_rate(stm->output_desc.mSampleRate)) {
+    /* For uncommon sample rates, we may run into issues with the OS
+       resampler if we don't do the resampling ourselves, so set the
+       AudioUnit sample rate to the hardware rate and resample. */
+    stm->output_desc.mSampleRate = stm->output_hw_rate;
+  }
   LOG("(%p) Output device sampling rate: %.2f", stm,
       output_hw_desc.mSampleRate);
   stm->context->channels = output_hw_desc.mChannelsPerFrame;
@@ -2709,11 +2724,16 @@ audiounit_setup_stream(cubeb_stream * stm)
     input_unconverted_params.rate = stm->input_hw_rate;
   }
 
-  /* Create resampler. Output params are unchanged
+  cubeb_stream_params output_unconverted_params;
-   * because we do not need conversion on the output. */
+  if (has_output(stm)) {
+    output_unconverted_params = stm->output_stream_params;
+    output_unconverted_params.rate = stm->output_desc.mSampleRate;
+  }
+
+  /* Create resampler. */
   stm->resampler.reset(cubeb_resampler_create(
       stm, has_input(stm) ? &input_unconverted_params : NULL,
-      has_output(stm) ? &stm->output_stream_params : NULL, target_sample_rate,
+      has_output(stm) ? &output_unconverted_params : NULL, target_sample_rate,
       stm->data_callback, stm->user_ptr, CUBEB_RESAMPLER_QUALITY_DESKTOP,
       CUBEB_RESAMPLER_RECLOCK_NONE));
   if (!stm->resampler) {
@@ -3665,6 +3685,7 @@ cubeb_ops const audiounit_ops = {
     /*.get_max_channel_count =*/audiounit_get_max_channel_count,
     /*.get_min_latency =*/audiounit_get_min_latency,
     /*.get_preferred_sample_rate =*/audiounit_get_preferred_sample_rate,
+    /*.get_supported_input_processing_params =*/NULL,
     /*.enumerate_devices =*/audiounit_enumerate_devices,
     /*.device_collection_destroy =*/audiounit_device_collection_destroy,
     /*.destroy =*/audiounit_destroy,
@@ -3678,6 +3699,8 @@ cubeb_ops const audiounit_ops = {
     /*.stream_set_volume =*/audiounit_stream_set_volume,
     /*.stream_set_name =*/NULL,
     /*.stream_get_current_device =*/audiounit_stream_get_current_device,
+    /*.stream_set_input_mute =*/NULL,
+    /*.stream_set_input_processing_params =*/NULL,
     /*.stream_device_destroy =*/audiounit_stream_device_destroy,
     /*.stream_register_device_changed_callback =*/
     audiounit_stream_register_device_changed_callback,

3rdparty/cubeb/src/cubeb_jack.cpp

@@ -160,6 +160,7 @@ static struct cubeb_ops const cbjack_ops = {
     .get_max_channel_count = cbjack_get_max_channel_count,
     .get_min_latency = cbjack_get_min_latency,
     .get_preferred_sample_rate = cbjack_get_preferred_sample_rate,
+    .get_supported_input_processing_params = NULL,
     .enumerate_devices = cbjack_enumerate_devices,
     .device_collection_destroy = cbjack_device_collection_destroy,
     .destroy = cbjack_destroy,
@@ -173,6 +174,8 @@ static struct cubeb_ops const cbjack_ops = {
     .stream_set_volume = cbjack_stream_set_volume,
     .stream_set_name = NULL,
     .stream_get_current_device = cbjack_stream_get_current_device,
+    .stream_set_input_mute = NULL,
+    .stream_set_input_processing_params = NULL,
     .stream_device_destroy = cbjack_stream_device_destroy,
     .stream_register_device_changed_callback = NULL,
     .register_device_collection_changed = NULL};
@@ -431,8 +434,10 @@ cbjack_process(jack_nframes_t nframes, void * arg)
       if (stm->devs & OUT_ONLY) {
         for (unsigned int c = 0; c < stm->out_params.channels; c++) {
           float * buffer_out = bufs_out[c];
-          for (long f = 0; f < nframes; f++) {
+          if (buffer_out) {
-            buffer_out[f] = 0.f;
+            for (long f = 0; f < nframes; f++) {
+              buffer_out[f] = 0.f;
+            }
           }
         }
       }
@@ -440,8 +445,10 @@ cbjack_process(jack_nframes_t nframes, void * arg)
         // paused, capture silence
         for (unsigned int c = 0; c < stm->in_params.channels; c++) {
           float * buffer_in = bufs_in[c];
-          for (long f = 0; f < nframes; f++) {
+          if (buffer_in) {
-            buffer_in[f] = 0.f;
+            for (long f = 0; f < nframes; f++) {
+              buffer_in[f] = 0.f;
+            }
           }
         }
       }
@@ -493,8 +500,10 @@ cbjack_process(jack_nframes_t nframes, void * arg)
         if (stm->devs & OUT_ONLY) {
           for (unsigned int c = 0; c < stm->out_params.channels; c++) {
             float * buffer_out = bufs_out[c];
-            for (long f = 0; f < nframes; f++) {
+            if (buffer_out) {
-              buffer_out[f] = 0.f;
+              for (long f = 0; f < nframes; f++) {
+                buffer_out[f] = 0.f;
+              }
             }
           }
         }
@@ -502,8 +511,10 @@ cbjack_process(jack_nframes_t nframes, void * arg)
           // capture silence
           for (unsigned int c = 0; c < stm->in_params.channels; c++) {
             float * buffer_in = bufs_in[c];
-            for (long f = 0; f < nframes; f++) {
+            if (buffer_in) {
-              buffer_in[f] = 0.f;
+              for (long f = 0; f < nframes; f++) {
+                buffer_in[f] = 0.f;
+              }
             }
           }
         }
@@ -542,20 +553,26 @@ cbjack_deinterleave_playback_refill_float(cubeb_stream * stream, float ** in,
     for (unsigned int c = 0; c < stream->out_params.channels; c++) {
       float * buffer = bufs_out[c];
       for (long f = 0; f < done_frames; f++) {
-        buffer[f] =
+        if (buffer) {
-            out_interleaved_buffer[(f * stream->out_params.channels) + c] *
+          buffer[f] =
-            stream->volume;
+              out_interleaved_buffer[(f * stream->out_params.channels) + c] *
+              stream->volume;
+        }
       }
       if (done_frames < needed_frames) {
         // draining
         for (long f = done_frames; f < needed_frames; f++) {
-          buffer[f] = 0.f;
+          if (buffer) {
+            buffer[f] = 0.f;
+          }
         }
       }
       if (done_frames == 0) {
         // stop, but first zero out the existing buffer
         for (long f = 0; f < needed_frames; f++) {
-          buffer[f] = 0.f;
+          if (buffer) {
+            buffer[f] = 0.f;
+          }
         }
       }
     }

3rdparty/cubeb/src/cubeb_oss.c

@@ -1335,6 +1335,7 @@ static struct cubeb_ops const oss_ops = {
     .get_max_channel_count = oss_get_max_channel_count,
     .get_min_latency = oss_get_min_latency,
     .get_preferred_sample_rate = oss_get_preferred_sample_rate,
+    .get_supported_input_processing_params = NULL,
     .enumerate_devices = oss_enumerate_devices,
     .device_collection_destroy = oss_device_collection_destroy,
     .destroy = oss_destroy,
@@ -1348,6 +1349,8 @@ static struct cubeb_ops const oss_ops = {
     .stream_set_volume = oss_stream_set_volume,
     .stream_set_name = NULL,
     .stream_get_current_device = oss_get_current_device,
+    .stream_set_input_mute = NULL,
+    .stream_set_input_processing_params = NULL,
     .stream_device_destroy = oss_stream_device_destroy,
     .stream_register_device_changed_callback = NULL,
     .register_device_collection_changed = NULL};

3rdparty/cubeb/src/cubeb_pulse.c

@@ -1690,6 +1690,7 @@ static struct cubeb_ops const pulse_ops = {
     .get_max_channel_count = pulse_get_max_channel_count,
     .get_min_latency = pulse_get_min_latency,
     .get_preferred_sample_rate = pulse_get_preferred_sample_rate,
+    .get_supported_input_processing_params = NULL,
     .enumerate_devices = pulse_enumerate_devices,
     .device_collection_destroy = pulse_device_collection_destroy,
     .destroy = pulse_destroy,
@@ -1703,6 +1704,8 @@ static struct cubeb_ops const pulse_ops = {
     .stream_set_volume = pulse_stream_set_volume,
     .stream_set_name = pulse_stream_set_name,
     .stream_get_current_device = pulse_stream_get_current_device,
+    .stream_set_input_mute = NULL,
+    .stream_set_input_processing_params = NULL,
     .stream_device_destroy = pulse_stream_device_destroy,
     .stream_register_device_changed_callback = NULL,
     .register_device_collection_changed =

3rdparty/cubeb/src/cubeb_sndio.c

@@ -667,6 +667,7 @@ static struct cubeb_ops const sndio_ops = {
     .get_max_channel_count = sndio_get_max_channel_count,
     .get_min_latency = sndio_get_min_latency,
     .get_preferred_sample_rate = sndio_get_preferred_sample_rate,
+    .get_supported_input_processing_params = NULL,
     .enumerate_devices = sndio_enumerate_devices,
     .device_collection_destroy = sndio_device_collection_destroy,
     .destroy = sndio_destroy,
@@ -679,6 +680,8 @@ static struct cubeb_ops const sndio_ops = {
     .stream_set_volume = sndio_stream_set_volume,
     .stream_set_name = NULL,
     .stream_get_current_device = NULL,
+    .stream_set_input_mute = NULL,
+    .stream_set_input_processing_params = NULL,
     .stream_device_destroy = NULL,
     .stream_register_device_changed_callback = NULL,
     .register_device_collection_changed = NULL};

3rdparty/cubeb/src/cubeb_triple_buffer.h

@@ -42,6 +42,13 @@ public:
   {
     return (shared_state.load(std::memory_order_relaxed) & BACK_DIRTY_BIT) != 0;
   }
+  // Reset state and indices to initial values.
+  void invalidate()
+  {
+    shared_state.store(0, std::memory_order_release);
+    input_idx = 1;
+    output_idx = 2;
+  }
 
 private:
   // Publish a value to the consumer. Returns true if the data was overwritten

3rdparty/cubeb/src/cubeb_utils.h

@@ -182,7 +182,9 @@ public:
     if (length_ + length > capacity_) {
       reserve(length_ + length);
     }
-    PodCopy(data_ + length_, elements, length);
+    if (data_) {
+      PodCopy(data_ + length_, elements, length);
+    }
     length_ += length;
   }
 
@@ -195,12 +197,14 @@ public:
     if (length_ + length > capacity_) {
       reserve(length + length_);
     }
-    PodZero(data_ + length_, length);
+    if (data_) {
+      PodZero(data_ + length_, length);
+    }
     length_ += length;
   }
 
-  /** Prepend `length` zero-ed elements to the end of the array, resizing the
+  /** Prepend `length` zero-ed elements to the front of the array, resizing and
-   * array if needed.
+   * shifting the array if needed.
    * @parameter length the number of elements to prepend to the array.
    */
   void push_front_silence(size_t length)
@@ -208,8 +212,10 @@ public:
     if (length_ + length > capacity_) {
       reserve(length + length_);
     }
-    PodMove(data_ + length, data_, length_);
+    if (data_) {
-    PodZero(data_, length);
+      PodMove(data_ + length, data_, length_);
+      PodZero(data_, length);
+    }
     length_ += length;
   }
 
@@ -227,6 +233,9 @@ public:
     if (length > length_) {
       return false;
     }
+    if (!data_) {
+      return true;
+    }
     if (elements) {
       PodCopy(elements, data_, length);
     }

3rdparty/cubeb/src/cubeb_wasapi.cpp

@@ -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
@@ -204,6 +229,11 @@ struct auto_stream_ref {
   cubeb_stream * stm;
 };
 
+using set_mm_thread_characteristics_function =
+    decltype(&AvSetMmThreadCharacteristicsW);
+using revert_mm_thread_characteristics_function =
+    decltype(&AvRevertMmThreadCharacteristics);
+
 extern cubeb_ops const wasapi_ops;
 
 static com_heap_ptr<wchar_t>
@@ -301,6 +331,13 @@ struct cubeb {
       nullptr;
   void * output_collection_changed_user_ptr = nullptr;
   UINT64 performance_counter_frequency;
+  /* Library dynamically opened to increase the render thread priority, and
+     the two function pointers we need. */
+  HMODULE mmcss_module = nullptr;
+  set_mm_thread_characteristics_function set_mm_thread_characteristics =
+      nullptr;
+  revert_mm_thread_characteristics_function revert_mm_thread_characteristics =
+      nullptr;
 };
 
 class wasapi_endpoint_notification_client;
@@ -1350,10 +1387,8 @@ refill_callback_output(cubeb_stream * stm)
 
   long got = refill(stm, nullptr, 0, output_buffer, output_frames);
 
-  if (got != output_frames) {
+  ALOGV("Output callback: output frames requested: %Iu, got %ld", output_frames,
-    ALOGV("Output callback: output frames requested: %Iu, got %ld", output_frames,
+        got);
-          got);
-  }
   if (got < 0) {
     return false;
   }
@@ -1403,7 +1438,8 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)
 
   /* We could consider using "Pro Audio" here for WebAudio and
      maybe WebRTC. */
-  mmcss_handle = AvSetMmThreadCharacteristicsA("Audio", &mmcss_task_index);
+  mmcss_handle =
+      stm->context->set_mm_thread_characteristics(L"Audio", &mmcss_task_index);
   if (!mmcss_handle) {
     /* This is not fatal, but we might glitch under heavy load. */
     LOG("Unable to use mmcss to bump the render thread priority: %lx",
@@ -1511,7 +1547,7 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)
   }
 
   if (mmcss_handle) {
-    AvRevertMmThreadCharacteristics(mmcss_handle);
+    stm->context->revert_mm_thread_characteristics(mmcss_handle);
   }
 
   if (FAILED(hr)) {
@@ -1524,6 +1560,18 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)
 void
 wasapi_destroy(cubeb * context);
 
+HANDLE WINAPI
+set_mm_thread_characteristics_noop(LPCWSTR, LPDWORD mmcss_task_index)
+{
+  return (HANDLE)1;
+}
+
+BOOL WINAPI
+revert_mm_thread_characteristics_noop(HANDLE mmcss_handle)
+{
+  return true;
+}
+
 HRESULT
 register_notification_client(cubeb_stream * stm)
 {
@@ -1759,6 +1807,31 @@ wasapi_init(cubeb ** context, char const * context_name)
     ctx->performance_counter_frequency = 0;
   }
 
+  ctx->mmcss_module = LoadLibraryW(L"Avrt.dll");
+
+  bool success = false;
+  if (ctx->mmcss_module) {
+    ctx->set_mm_thread_characteristics =
+        reinterpret_cast<set_mm_thread_characteristics_function>(
+            GetProcAddress(ctx->mmcss_module, "AvSetMmThreadCharacteristicsW"));
+    ctx->revert_mm_thread_characteristics =
+        reinterpret_cast<revert_mm_thread_characteristics_function>(
+            GetProcAddress(ctx->mmcss_module,
+                           "AvRevertMmThreadCharacteristics"));
+    success = ctx->set_mm_thread_characteristics &&
+              ctx->revert_mm_thread_characteristics;
+  }
+  if (!success) {
+    // This is not a fatal error, but we might end up glitching when
+    // the system is under high load.
+    LOG("Could not load avrt.dll or fetch AvSetMmThreadCharacteristicsW "
+        "AvRevertMmThreadCharacteristics: %lx",
+        GetLastError());
+    ctx->set_mm_thread_characteristics = &set_mm_thread_characteristics_noop;
+    ctx->revert_mm_thread_characteristics =
+        &revert_mm_thread_characteristics_noop;
+  }
+
   *context = ctx;
 
   return CUBEB_OK;
@@ -1815,6 +1888,10 @@ wasapi_destroy(cubeb * context)
     }
   }
 
+  if (context->mmcss_module) {
+    FreeLibrary(context->mmcss_module);
+  }
+
   delete context;
 }
 
@@ -1872,6 +1949,44 @@ 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 = hns_to_frames(params.rate, min_period_rt);
+
+        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());
@@ -1891,18 +2006,8 @@ 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.
+  // The minimum_period is only relevant in exclusive streams.
-     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
   *latency_frames = hns_to_frames(params.rate, default_period);
-#endif
 
   LOG("Minimum latency in frames: %u", *latency_frames);
 
@@ -1992,7 +2097,10 @@ 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 =
@@ -2036,12 +2144,12 @@ 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());
@@ -2057,24 +2165,22 @@ 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
+  //   might have their latency adjusted to match. When the shared-mode stream
-  //   have their latency adjusted to match. When  the shared-mode stream is
+  //   is closed, they'll go back to normal.
-  //   closed, they'll go back to normal.
+  // - If there's already a shared-mode stream running, if it created with the
-  // - If there's already a shared-mode stream running, then we cannot request
+  //   AUDCLNT_STREAMOPTIONS_MATCH_FORMAT option, the audio engine would be
-  //   the engine change to a different latency - we have to match it.
+  //   locked to that format, so we have to match it (a custom one would fail).
-  // - It's antisocial to lock the WASAPI engine at its default latency. If we
+  // - We don't lock the WASAPI engine to a format, as it's antisocial towards
-  //   would do this, then stop and use IAudioClient instead.
+  //   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,
@@ -2086,28 +2192,59 @@ 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 "
+    LOG("Requested latency %i equal or greater than default latency %i,"
-        "IAudioClient3",
+        " 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) {
@@ -2119,23 +2256,38 @@ 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;
   }
-
+  com_heap_ptr<WAVEFORMATEX> current_format(current_format_ptr);
-  if (current_period >= default_period) {
+  if (current_format->nSamplesPerSec != mix_format->nSamplesPerSec) {
-    LOG("Current shared mode engine period %i too high, not using IAudioClient",
+    // Unless some other external app locked the shared mode engine period
-        current_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)) {
@@ -2147,7 +2299,6 @@ 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")
 
@@ -2171,6 +2322,12 @@ 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.
@@ -2207,17 +2364,14 @@ 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
+    if (allow_audio_client_3) {
-    hr = device->Activate(__uuidof(IAudioClient3),
+      hr = device->Activate(__uuidof(IAudioClient3), CLSCTX_INPROC_SERVER, NULL,
-                          CLSCTX_INPROC_SERVER,
+                            audio_client.receive_vpp());
-                          NULL, audio_client.receive_vpp());
+    }
-    if (hr == E_NOINTERFACE) {
+    if (!allow_audio_client_3 || hr == E_NOINTERFACE) {
-#endif
+      hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
-    hr = device->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL,
+                            audio_client.receive_vpp());
-                          audio_client.receive_vpp());
-#if 0
     }
-#endif
 
     if (FAILED(hr)) {
       LOG("Could not activate the device to get an audio"
@@ -2346,16 +2500,15 @@ setup_wasapi_stream_one_side(cubeb_stream * stm,
     }
   }
 
-#if 0 // See https://bugzilla.mozilla.org/show_bug.cgi?id=1590902
+  if (allow_audio_client_3 &&
-  if (initialize_iaudioclient3(audio_client, stm, mix_format, flags, direction)) {
+      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,
-  hr = audio_client->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, latency_hns, 0,
+                                  0, mix_format.get(), NULL);
-                                mix_format.get(), NULL);
-#if 0
   }
-#endif
+
   if (FAILED(hr)) {
     LOG("Unable to initialize audio client for %s: %lx.", DIRECTION_NAME, hr);
     return CUBEB_ERROR;
@@ -3351,6 +3504,7 @@ 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)) {
@@ -3360,8 +3514,7 @@ 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 =
+      wfx = reinterpret_cast<WAVEFORMATEX *>(fmtvar.blob.pBlobData);
-          reinterpret_cast<WAVEFORMATEX *>(fmtvar.blob.pBlobData);
 
       if (fmtvar.blob.cbSize >= sizeof(WAVEFORMATEX) + wfx->cbSize ||
           wfx->wFormatTag == WAVE_FORMAT_PCM) {
@@ -3371,9 +3524,30 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
     }
   }
 
-  if (SUCCEEDED(dev->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER,
+#if USE_AUDIO_CLIENT_3_MIN_PERIOD
-                              NULL, client.receive_vpp())) &&
+  // Here we assume an IAudioClient3 stream will successfully
-      SUCCEEDED(client->GetDevicePeriod(&def_period, &min_period))) {
+  // 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 {
@@ -3562,6 +3736,7 @@ cubeb_ops const wasapi_ops = {
     /*.get_max_channel_count =*/wasapi_get_max_channel_count,
     /*.get_min_latency =*/wasapi_get_min_latency,
     /*.get_preferred_sample_rate =*/wasapi_get_preferred_sample_rate,
+    /*.get_supported_input_processing_params =*/NULL,
     /*.enumerate_devices =*/wasapi_enumerate_devices,
     /*.device_collection_destroy =*/wasapi_device_collection_destroy,
     /*.destroy =*/wasapi_destroy,
@@ -3575,6 +3750,8 @@ cubeb_ops const wasapi_ops = {
     /*.stream_set_volume =*/wasapi_stream_set_volume,
     /*.stream_set_name =*/NULL,
     /*.stream_get_current_device =*/NULL,
+    /*.stream_set_input_mute =*/NULL,
+    /*.stream_set_input_processing_params =*/NULL,
     /*.stream_device_destroy =*/NULL,
     /*.stream_register_device_changed_callback =*/NULL,
     /*.register_device_collection_changed =*/

3rdparty/cubeb/src/cubeb_winmm.c

@@ -1192,6 +1192,7 @@ static struct cubeb_ops const winmm_ops = {
     /*.get_max_channel_count=*/winmm_get_max_channel_count,
     /*.get_min_latency=*/winmm_get_min_latency,
     /*.get_preferred_sample_rate =*/winmm_get_preferred_sample_rate,
+    /*.get_supported_input_processing_params =*/NULL,
     /*.enumerate_devices =*/winmm_enumerate_devices,
     /*.device_collection_destroy =*/winmm_device_collection_destroy,
     /*.destroy =*/winmm_destroy,
@@ -1205,6 +1206,8 @@ static struct cubeb_ops const winmm_ops = {
     /*.stream_set_volume =*/winmm_stream_set_volume,
     /*.stream_set_name =*/NULL,
     /*.stream_get_current_device =*/NULL,
+    /*.stream_set_input_mute =*/NULL,
+    /*.stream_set_input_processing_params =*/NULL,
     /*.stream_device_destroy =*/NULL,
     /*.stream_register_device_changed_callback=*/NULL,
     /*.register_device_collection_changed =*/NULL};