3rdparty: Update cubeb to e495bee

2025-12-16 04:08:48 +00:00 · 2025-10-01 16:19:27 +01:00 · 2025-10-01 16:19:27 +01:00 · bc11ff0571
commit bc11ff0571
parent e550cf9b63
16 changed files with 960 additions and 477 deletions
--- a/3rdparty/cubeb/README.md
+++ b/3rdparty/cubeb/README.md
@ -1,7 +1,117 @@
+# libcubeb - Cross-platform Audio I/O Library
+
 [![Build Status](https://github.com/mozilla/cubeb/actions/workflows/build.yml/badge.svg)](https://github.com/mozilla/cubeb/actions/workflows/build.yml)

-See INSTALL.md for build instructions.
+`libcubeb` is a cross-platform C library for high and low-latency audio input/output. It provides a simple, consistent API for audio playback and recording across multiple platforms and audio backends. It is written in C, C++ and Rust, with a C ABI and [Rust](https://github.com/mozilla/cubeb-rs) bindings. While originally written for use in the Firefox Web browser, a number of other software projects have adopted it.

-See [Backend Support](https://github.com/mozilla/cubeb/wiki/Backend-Support) in the wiki for the support level of each backend.
+## Features

-Licensed under an ISC-style license.  See LICENSE for details.
+- **Cross-platform support**: Windows, macOS, Linux, Android, and other platforms
+- **Versatile**: Optimized for low-latency real-time audio applications, or power efficient higher latency playback
+- **A/V sync**: Latency compensated audio clock reporting for easy audio/video synchronization
+- **Full-duplex support**: Simultaneous audio input and output, reclocked
+- **Device enumeration**: Query available audio devices
+- **Audio processing for speech**: Can use VoiceProcessing IO on recent macOS
+
+## Supported Backends & status
+
+| *Backend*         | *Support Level* | *Platform version* | *Notes*                                          |
+|-------------------|-----------------|--------------------|--------------------------------------------------|
+| PulseAudio (Rust) | Tier-1          |                    | Main Linux desktop backend                       |
+| AudioUnit (Rust)  | Tier-1          |                    | Main macOS backend                               |
+| WASAPI            | Tier-1          | Windows >= 7       | Main Windows backend                             |
+| AAudio            | Tier-1          | Android >= 8       | Main Android backend for most devices            |
+| OpenSL            | Tier-1          | Android >= 2.3     | Android backend for older devices                |
+| OSS               | Tier-2          |                    |                                                  |
+| sndio             | Tier-2          |                    |                                                  |
+| Sun               | Tier-2          |                    |                                                  |
+| WinMM             | Tier-3          | Windows XP         | Was Tier-1, Firefox minimum Windows version 7.   |
+| AudioTrack        | Tier-3          | Android < 2.3      | Was Tier-1, Firefox minimum Android version 4.1. |
+| ALSA              | Tier-3          |                    |                                                  |
+| JACK              | Tier-3          |                    |                                                  |
+| KAI               | Tier-3          |                    |                                                  |
+| PulseAudio (C)    | Tier-4          |                    | Was Tier-1, superseded by Rust                   |
+| AudioUnit (C++)   | Tier-4          |                    | Was Tier-1, superseded by Rust                   |
+
+Tier-1: Actively maintained.  Should have CI coverage. Critical for Firefox.
+
+Tier-2: Actively maintained by contributors.  CI coverage appreciated.
+
+Tier-3: Maintainers/patches accepted.  Status unclear.
+
+Tier-4: Deprecated, obsolete.  Scheduled to be removed.
+
+Note that the support level is not a judgement of the relative merits
+of a backend, only the current state of support, which is informed
+by Firefox's needs, the responsiveness of a backend's
+maintainer, and the level of contributions to that backend.
+
+## Building
+
+### Prerequisites
+
+- CMake 3.15 or later
+- Non-ancient MSVC, clang or gcc, for compiling both C and C++
+- Platform-specific audio libraries (automatically detected)
+- Optional but recommended: Rust compiler to compile and link more recent backends for macOS and PulseAudio
+
+### Quick build
+
+```bash
+git clone https://github.com/mozilla/cubeb.git
+cd cubeb
+cmake -B build
+cmake --build build
+```
+
+### Better build with Rust backends
+
+```bash
+git clone --recursive https://github.com/mozilla/cubeb.git
+cd cubeb
+cmake -B build -DBUILD_RUST_LIBS=ON
+cmake --build build
+```
+
+### Platform-Specific Notes
+
+**Windows**: Supports Visual Studio 2015+ and MinGW-w64. Use `-G "Visual Studio 16 2019"` or `-G "MinGW Makefiles"`.
+
+**macOS**: Requires Xcode command line tools. Audio frameworks are automatically linked.
+
+**Linux**: Development packages for desired backends:
+```bash
+# Ubuntu/Debian
+sudo apt-get install libpulse-dev libasound2-dev libjack-dev
+
+# Fedora/RHEL
+sudo dnf install pulseaudio-libs-devel alsa-lib-devel jack-audio-connection-kit-devel
+```
+
+**Android**: Use with Android NDK. AAudio requires API level 26+.
+
+## Testing
+
+Run the test suite:
+```bash
+cd build
+ctest
+```
+
+Use the interactive test tool:
+```bash
+./cubeb-test
+```
+
+## License
+
+Licensed under an ISC-style license. See [LICENSE](LICENSE) for details.
+
+## Contributing
+
+Contributions are welcome! Please see the [contribution guidelines](CONTRIBUTING.md) and check the [issue tracker](https://github.com/mozilla/cubeb/issues).
+
+## Links
+
+- [GitHub Repository](https://github.com/mozilla/cubeb)
+- [API Documentation](https://mozilla.github.io/cubeb/)
--- a/3rdparty/cubeb/include/cubeb/cubeb.h
+++ b/3rdparty/cubeb/include/cubeb/cubeb.h
@ -49,6 +49,7 @@ extern "C" {
    output_params.channels = 2;
    output_params.layout = CUBEB_LAYOUT_UNDEFINED;
    output_params.prefs = CUBEB_STREAM_PREF_NONE;
+    output_params.input_params = CUBEB_INPUT_PROCESSING_PARAM_NONE;

    rv = cubeb_get_min_latency(app_ctx, &output_params, &latency_frames);
    if (rv != CUBEB_OK) {
@ -62,6 +63,7 @@ extern "C" {
    input_params.channels = 1;
    input_params.layout = CUBEB_LAYOUT_UNDEFINED;
    input_params.prefs = CUBEB_STREAM_PREF_NONE;
+    input_params.input_params = CUBEB_INPUT_PROCESSING_PARAM_NONE;

    cubeb_stream * stm;
    rv = cubeb_stream_init(app_ctx, &stm, "Example Stream 1",
@ -193,39 +195,39 @@ typedef uint32_t cubeb_channel_layout;
 // Some common layout definitions.
 enum {
  CUBEB_LAYOUT_UNDEFINED = 0, // Indicate the speaker's layout is undefined.
-  CUBEB_LAYOUT_MONO = (uint32_t)CHANNEL_FRONT_CENTER,
-  CUBEB_LAYOUT_MONO_LFE = (uint32_t)CUBEB_LAYOUT_MONO | (uint32_t)CHANNEL_LOW_FREQUENCY,
-  CUBEB_LAYOUT_STEREO = (uint32_t)CHANNEL_FRONT_LEFT | (uint32_t)CHANNEL_FRONT_RIGHT,
-  CUBEB_LAYOUT_STEREO_LFE = (uint32_t)CUBEB_LAYOUT_STEREO | (uint32_t)CHANNEL_LOW_FREQUENCY,
+  CUBEB_LAYOUT_MONO = CHANNEL_FRONT_CENTER,
+  CUBEB_LAYOUT_MONO_LFE = CUBEB_LAYOUT_MONO | CHANNEL_LOW_FREQUENCY,
+  CUBEB_LAYOUT_STEREO = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT,
+  CUBEB_LAYOUT_STEREO_LFE = CUBEB_LAYOUT_STEREO | CHANNEL_LOW_FREQUENCY,
  CUBEB_LAYOUT_3F =
-      (uint32_t)CHANNEL_FRONT_LEFT | (uint32_t)CHANNEL_FRONT_RIGHT | (uint32_t)CHANNEL_FRONT_CENTER,
-  CUBEB_LAYOUT_3F_LFE = (uint32_t)CUBEB_LAYOUT_3F | (uint32_t)CHANNEL_LOW_FREQUENCY,
+      CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_FRONT_CENTER,
+  CUBEB_LAYOUT_3F_LFE = CUBEB_LAYOUT_3F | CHANNEL_LOW_FREQUENCY,
  CUBEB_LAYOUT_2F1 =
-      (uint32_t)CHANNEL_FRONT_LEFT | (uint32_t)CHANNEL_FRONT_RIGHT | (uint32_t)CHANNEL_BACK_CENTER,
-  CUBEB_LAYOUT_2F1_LFE = (uint32_t)CUBEB_LAYOUT_2F1 | (uint32_t)CHANNEL_LOW_FREQUENCY,
-  CUBEB_LAYOUT_3F1 = (uint32_t)CHANNEL_FRONT_LEFT | (uint32_t)CHANNEL_FRONT_RIGHT |
-                     (uint32_t)CHANNEL_FRONT_CENTER | (uint32_t)CHANNEL_BACK_CENTER,
-  CUBEB_LAYOUT_3F1_LFE = (uint32_t)CUBEB_LAYOUT_3F1 | (uint32_t)CHANNEL_LOW_FREQUENCY,
-  CUBEB_LAYOUT_2F2 = (uint32_t)CHANNEL_FRONT_LEFT | (uint32_t)CHANNEL_FRONT_RIGHT |
-                     (uint32_t)CHANNEL_SIDE_LEFT | (uint32_t)CHANNEL_SIDE_RIGHT,
-  CUBEB_LAYOUT_2F2_LFE = (uint32_t)CUBEB_LAYOUT_2F2 | (uint32_t)CHANNEL_LOW_FREQUENCY,
-  CUBEB_LAYOUT_QUAD = (uint32_t)CHANNEL_FRONT_LEFT | (uint32_t)CHANNEL_FRONT_RIGHT |
-                      (uint32_t)CHANNEL_BACK_LEFT | (uint32_t)CHANNEL_BACK_RIGHT,
-  CUBEB_LAYOUT_QUAD_LFE = (uint32_t)CUBEB_LAYOUT_QUAD | (uint32_t)CHANNEL_LOW_FREQUENCY,
-  CUBEB_LAYOUT_3F2 = (uint32_t)CHANNEL_FRONT_LEFT | (uint32_t)CHANNEL_FRONT_RIGHT |
-                     (uint32_t)CHANNEL_FRONT_CENTER | (uint32_t)CHANNEL_SIDE_LEFT |
-                     (uint32_t)CHANNEL_SIDE_RIGHT,
-  CUBEB_LAYOUT_3F2_LFE = (uint32_t)CUBEB_LAYOUT_3F2 | (uint32_t)CHANNEL_LOW_FREQUENCY,
-  CUBEB_LAYOUT_3F2_BACK = (uint32_t)CUBEB_LAYOUT_QUAD | (uint32_t)CHANNEL_FRONT_CENTER,
-  CUBEB_LAYOUT_3F2_LFE_BACK = (uint32_t)CUBEB_LAYOUT_3F2_BACK | (uint32_t)CHANNEL_LOW_FREQUENCY,
-  CUBEB_LAYOUT_3F3R_LFE = (uint32_t)CHANNEL_FRONT_LEFT | (uint32_t)CHANNEL_FRONT_RIGHT |
-                          (uint32_t)CHANNEL_FRONT_CENTER | (uint32_t)CHANNEL_LOW_FREQUENCY |
-                          (uint32_t)CHANNEL_BACK_CENTER | (uint32_t)CHANNEL_SIDE_LEFT |
-                          (uint32_t)CHANNEL_SIDE_RIGHT,
-  CUBEB_LAYOUT_3F4_LFE = (uint32_t)CHANNEL_FRONT_LEFT | (uint32_t)CHANNEL_FRONT_RIGHT |
-                         (uint32_t)CHANNEL_FRONT_CENTER | (uint32_t)CHANNEL_LOW_FREQUENCY |
-                         (uint32_t)CHANNEL_BACK_LEFT | (uint32_t)CHANNEL_BACK_RIGHT |
-                         (uint32_t)CHANNEL_SIDE_LEFT | (uint32_t)CHANNEL_SIDE_RIGHT,
+      CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_BACK_CENTER,
+  CUBEB_LAYOUT_2F1_LFE = CUBEB_LAYOUT_2F1 | CHANNEL_LOW_FREQUENCY,
+  CUBEB_LAYOUT_3F1 = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+                     CHANNEL_FRONT_CENTER | CHANNEL_BACK_CENTER,
+  CUBEB_LAYOUT_3F1_LFE = CUBEB_LAYOUT_3F1 | CHANNEL_LOW_FREQUENCY,
+  CUBEB_LAYOUT_2F2 = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+                     CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT,
+  CUBEB_LAYOUT_2F2_LFE = CUBEB_LAYOUT_2F2 | CHANNEL_LOW_FREQUENCY,
+  CUBEB_LAYOUT_QUAD = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+                      CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT,
+  CUBEB_LAYOUT_QUAD_LFE = CUBEB_LAYOUT_QUAD | CHANNEL_LOW_FREQUENCY,
+  CUBEB_LAYOUT_3F2 = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+                     CHANNEL_FRONT_CENTER | CHANNEL_SIDE_LEFT |
+                     CHANNEL_SIDE_RIGHT,
+  CUBEB_LAYOUT_3F2_LFE = CUBEB_LAYOUT_3F2 | CHANNEL_LOW_FREQUENCY,
+  CUBEB_LAYOUT_3F2_BACK = CUBEB_LAYOUT_QUAD | CHANNEL_FRONT_CENTER,
+  CUBEB_LAYOUT_3F2_LFE_BACK = CUBEB_LAYOUT_3F2_BACK | CHANNEL_LOW_FREQUENCY,
+  CUBEB_LAYOUT_3F3R_LFE = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+                          CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY |
+                          CHANNEL_BACK_CENTER | CHANNEL_SIDE_LEFT |
+                          CHANNEL_SIDE_RIGHT,
+  CUBEB_LAYOUT_3F4_LFE = CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+                         CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY |
+                         CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT |
+                         CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT,
 };

 /** Miscellaneous stream preferences. */
@ -280,6 +282,9 @@ typedef struct {
      layout; /**< Requested channel layout. This must be consistent with the
                 provided channels. CUBEB_LAYOUT_UNDEFINED if unknown */
  cubeb_stream_prefs prefs;                   /**< Requested preferences. */
+  cubeb_input_processing_params input_params; /**< Requested input processing
+     params. Ignored for output streams. At present, only supported on the
+     WASAPI backend; others should use cubeb_set_input_processing_params.  */
 } cubeb_stream_params;

 /** Audio device description */
@ -414,6 +419,13 @@ typedef struct {
  size_t count;               /**< Device count in collection. */
 } cubeb_device_collection;

+/** Array of compiled backends returned by `cubeb_get_backend_names`. */
+typedef struct {
+  const char * const *
+      names;    /**< Array of strings representing backend names. */
+  size_t count; /**< Length of the array. */
+} cubeb_backend_names;
+
 /** User supplied data callback.
    - Calling other cubeb functions from this callback is unsafe.
    - The code in the callback should be non-blocking.
@ -454,6 +466,8 @@ typedef void (*cubeb_device_changed_callback)(void * user_ptr);

 /**
 * User supplied callback called when the underlying device collection changed.
+ * This callback will be called when devices are added or removed from the
+ * system, or when the default device changes for the specified device type.
 * @param context A pointer to the cubeb context.
 * @param user_ptr The pointer passed to
 * cubeb_register_device_collection_changed. */
@ -485,17 +499,18 @@ CUBEB_EXPORT int
 cubeb_init(cubeb ** context, char const * context_name,
           char const * backend_name);

-/** Returns a list of backend names which can be supplid to cubeb_init().
-    Array is null-terminated. */
-CUBEB_EXPORT const char**
-cubeb_get_backend_names();
-
 /** Get a read-only string identifying this context's current backend.
    @param context A pointer to the cubeb context.
    @retval Read-only string identifying current backend. */
 CUBEB_EXPORT char const *
 cubeb_get_backend_id(cubeb * context);

+/** Get a read-only array of strings identifying available backends.
+    These can be passed as `backend_name` parameter to `cubeb_init`.
+    @retval Struct containing the array with backend names. */
+CUBEB_EXPORT cubeb_backend_names
+cubeb_get_backend_names();
+
 /** Get the maximum possible number of channels.
    @param context A pointer to the cubeb context.
    @param max_channels The maximum number of channels.
@ -674,7 +689,7 @@ cubeb_stream_get_current_device(cubeb_stream * stm,
    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
+    @param mute whether the input should mute or not
    @retval CUBEB_OK
    @retval CUBEB_ERROR_INVALID_PARAMETER if this stream does not have an input
            device
@ -745,14 +760,16 @@ cubeb_device_collection_destroy(cubeb * context,
                                cubeb_device_collection * collection);

 /** Registers a callback which is called when the system detects
-    a new device or a device is removed.
+    a new device or a device is removed, or when the default device
+    changes for the specified device type.
    @param context
    @param devtype device type to include. Different callbacks and user pointers
           can be registered for each devtype. The hybrid devtype
           `CUBEB_DEVICE_TYPE_INPUT | CUBEB_DEVICE_TYPE_OUTPUT` is also valid
           and will register the provided callback and user pointer in both
   sides.
-    @param callback a function called whenever the system device list changes.
+    @param callback a function called whenever the system device list changes,
+           including when default devices change.
           Passing NULL allow to unregister a function. You have to unregister
           first before you register a new callback.
    @param user_ptr pointer to user specified data which will be present in
--- a/3rdparty/cubeb/src/cubeb.c
+++ b/3rdparty/cubeb/src/cubeb.c
@ -31,6 +31,10 @@ struct cubeb_stream {
 int
 pulse_init(cubeb ** context, char const * context_name);
 #endif
+#if defined(USE_PULSE_RUST)
+int
+pulse_rust_init(cubeb ** contet, char const * context_name);
+#endif
 #if defined(USE_JACK)
 int
 jack_init(cubeb ** context, char const * context_name);
@ -43,6 +47,10 @@ alsa_init(cubeb ** context, char const * context_name);
 int
 audiounit_init(cubeb ** context, char const * context_name);
 #endif
+#if defined(USE_AUDIOUNIT_RUST)
+int
+audiounit_rust_init(cubeb ** contet, char const * context_name);
+#endif
 #if defined(USE_WINMM)
 int
 winmm_init(cubeb ** context, char const * context_name);
@ -55,10 +63,30 @@ wasapi_init(cubeb ** context, char const * context_name);
 int
 sndio_init(cubeb ** context, char const * context_name);
 #endif
+#if defined(USE_SUN)
+int
+sun_init(cubeb ** context, char const * context_name);
+#endif
+#if defined(USE_OPENSL)
+int
+opensl_init(cubeb ** context, char const * context_name);
+#endif
 #if defined(USE_OSS)
 int
 oss_init(cubeb ** context, char const * context_name);
 #endif
+#if defined(USE_AAUDIO)
+int
+aaudio_init(cubeb ** context, char const * context_name);
+#endif
+#if defined(USE_AUDIOTRACK)
+int
+audiotrack_init(cubeb ** context, char const * context_name);
+#endif
+#if defined(USE_KAI)
+int
+kai_init(cubeb ** context, char const * context_name);
+#endif

 static int
 validate_stream_params(cubeb_stream_params * input_stream_params,
@ -123,6 +151,10 @@ cubeb_init(cubeb ** context, char const * context_name,
    if (!strcmp(backend_name, "pulse")) {
 #if defined(USE_PULSE)
      init_oneshot = pulse_init;
+#endif
+    } else if (!strcmp(backend_name, "pulse-rust")) {
+#if defined(USE_PULSE_RUST)
+      init_oneshot = pulse_rust_init;
 #endif
    } else if (!strcmp(backend_name, "jack")) {
 #if defined(USE_JACK)
@ -135,6 +167,10 @@ cubeb_init(cubeb ** context, char const * context_name,
    } else if (!strcmp(backend_name, "audiounit")) {
 #if defined(USE_AUDIOUNIT)
      init_oneshot = audiounit_init;
+#endif
+    } else if (!strcmp(backend_name, "audiounit-rust")) {
+#if defined(USE_AUDIOUNIT_RUST)
+      init_oneshot = audiounit_rust_init;
 #endif
    } else if (!strcmp(backend_name, "wasapi")) {
 #if defined(USE_WASAPI)
@ -147,10 +183,30 @@ cubeb_init(cubeb ** context, char const * context_name,
    } else if (!strcmp(backend_name, "sndio")) {
 #if defined(USE_SNDIO)
      init_oneshot = sndio_init;
+#endif
+    } else if (!strcmp(backend_name, "sun")) {
+#if defined(USE_SUN)
+      init_oneshot = sun_init;
+#endif
+    } else if (!strcmp(backend_name, "opensl")) {
+#if defined(USE_OPENSL)
+      init_oneshot = opensl_init;
 #endif
    } else if (!strcmp(backend_name, "oss")) {
 #if defined(USE_OSS)
      init_oneshot = oss_init;
+#endif
+    } else if (!strcmp(backend_name, "aaudio")) {
+#if defined(USE_AAUDIO)
+      init_oneshot = aaudio_init;
+#endif
+    } else if (!strcmp(backend_name, "audiotrack")) {
+#if defined(USE_AUDIOTRACK)
+      init_oneshot = audiotrack_init;
+#endif
+    } else if (!strcmp(backend_name, "kai")) {
+#if defined(USE_KAI)
+      init_oneshot = kai_init;
 #endif
    } else {
      /* Already set */
@ -163,6 +219,9 @@ cubeb_init(cubeb ** context, char const * context_name,
     * to override all other choices
     */
    init_oneshot,
+#if defined(USE_PULSE_RUST)
+    pulse_rust_init,
+#endif
 #if defined(USE_PULSE)
    pulse_init,
 #endif
@ -178,6 +237,9 @@ cubeb_init(cubeb ** context, char const * context_name,
 #if defined(USE_OSS)
    oss_init,
 #endif
+#if defined(USE_AUDIOUNIT_RUST)
+    audiounit_rust_init,
+#endif
 #if defined(USE_AUDIOUNIT)
    audiounit_init,
 #endif
@ -189,6 +251,18 @@ cubeb_init(cubeb ** context, char const * context_name,
 #endif
 #if defined(USE_SUN)
    sun_init,
+#endif
+#if defined(USE_AAUDIO)
+    aaudio_init,
+#endif
+#if defined(USE_OPENSL)
+    opensl_init,
+#endif
+#if defined(USE_AUDIOTRACK)
+    audiotrack_init,
+#endif
+#if defined(USE_KAI)
+    kai_init,
 #endif
  };
  int i;
@ -214,13 +288,26 @@ cubeb_init(cubeb ** context, char const * context_name,
  return CUBEB_ERROR;
 }

-const char**
+char const *
+cubeb_get_backend_id(cubeb * context)
+{
+  if (!context) {
+    return NULL;
+  }
+
+  return context->ops->get_backend_id(context);
+}
+
+cubeb_backend_names
 cubeb_get_backend_names()
 {
-  static const char* backend_names[] = {
+  static const char * const backend_names[] = {
 #if defined(USE_PULSE)
    "pulse",
 #endif
+#if defined(USE_PULSE_RUST)
+    "pulse-rust",
+#endif
 #if defined(USE_JACK)
    "jack",
 #endif
@ -230,6 +317,9 @@ cubeb_get_backend_names()
 #if defined(USE_AUDIOUNIT)
    "audiounit",
 #endif
+#if defined(USE_AUDIOUNIT_RUST)
+    "audiounit-rust",
+#endif
 #if defined(USE_WASAPI)
    "wasapi",
 #endif
@ -239,23 +329,30 @@ cubeb_get_backend_names()
 #if defined(USE_SNDIO)
    "sndio",
 #endif
+#if defined(USE_SUN)
+    "sun",
+#endif
+#if defined(USE_OPENSL)
+    "opensl",
+#endif
 #if defined(USE_OSS)
    "oss",
 #endif
-    NULL,
+#if defined(USE_AAUDIO)
+    "aaudio",
+#endif
+#if defined(USE_AUDIOTRACK)
+    "audiotrack",
+#endif
+#if defined(USE_KAI)
+    "kai",
+#endif
  };

-  return backend_names;
-}
-
-char const *
-cubeb_get_backend_id(cubeb * context)
-{
-  if (!context) {
-    return NULL;
-  }
-
-  return context->ops->get_backend_id(context);
+  return (cubeb_backend_names){
+      .names = backend_names,
+      .count = NELEMS(backend_names),
+  };
 }

 int
--- a/3rdparty/cubeb/src/cubeb_audiounit.cpp
+++ b/3rdparty/cubeb/src/cubeb_audiounit.cpp
@ -213,12 +213,19 @@ struct cubeb_stream {
  cubeb_device_changed_callback device_changed_callback = nullptr;
  owned_critical_section device_changed_callback_lock;
  /* Stream creation parameters */
-  cubeb_stream_params input_stream_params = {CUBEB_SAMPLE_FLOAT32NE, 0, 0,
+  cubeb_stream_params input_stream_params = {CUBEB_SAMPLE_FLOAT32NE,
+                                             0,
+                                             0,
                                             CUBEB_LAYOUT_UNDEFINED,
-                                             CUBEB_STREAM_PREF_NONE};
-  cubeb_stream_params output_stream_params = {CUBEB_SAMPLE_FLOAT32NE, 0, 0,
+                                             CUBEB_STREAM_PREF_NONE,
+                                             CUBEB_INPUT_PROCESSING_PARAM_NONE};
+  cubeb_stream_params output_stream_params = {
+      CUBEB_SAMPLE_FLOAT32NE,
+      0,
+      0,
      CUBEB_LAYOUT_UNDEFINED,
-                                              CUBEB_STREAM_PREF_NONE};
+      CUBEB_STREAM_PREF_NONE,
+      CUBEB_INPUT_PROCESSING_PARAM_NONE};
  device_info input_device;
  device_info output_device;
  /* Format descriptions */
--- a/3rdparty/cubeb/src/cubeb_log.cpp
+++ b/3rdparty/cubeb/src/cubeb_log.cpp
@ -16,8 +16,8 @@
 #include <time.h>
 #endif

-static std::atomic<cubeb_log_level> g_cubeb_log_level;
-static std::atomic<cubeb_log_callback> g_cubeb_log_callback;
+std::atomic<cubeb_log_level> g_cubeb_log_level;
+std::atomic<cubeb_log_callback> g_cubeb_log_callback;

 /** The maximum size of a log message, after having been formatted. */
 const size_t CUBEB_LOG_MESSAGE_MAX_SIZE = 256;
@ -32,6 +32,133 @@ cubeb_noop_log_callback(char const * /* fmt */, ...)
 {
 }

+/**
+ * This wraps an inline buffer, that represents a log message, that must be
+ * null-terminated.
+ * This class should not use system calls or other potentially blocking code.
+ */
+class cubeb_log_message {
+public:
+  cubeb_log_message() { *storage = '\0'; }
+  cubeb_log_message(char const str[CUBEB_LOG_MESSAGE_MAX_SIZE])
+  {
+    size_t length = strlen(str);
+    /* paranoia against malformed message */
+    assert(length < CUBEB_LOG_MESSAGE_MAX_SIZE);
+    if (length > CUBEB_LOG_MESSAGE_MAX_SIZE - 1) {
+      return;
+    }
+    PodCopy(storage, str, length);
+    storage[length] = '\0';
+  }
+  char const * get() { return storage; }
+
+private:
+  char storage[CUBEB_LOG_MESSAGE_MAX_SIZE]{};
+};
+
+/** Lock-free asynchronous logger, made so that logging from a
+ *  real-time audio callback does not block the audio thread. */
+class cubeb_async_logger {
+public:
+  /* This is thread-safe since C++11 */
+  static cubeb_async_logger & get()
+  {
+    static cubeb_async_logger instance;
+    return instance;
+  }
+  void push(char const str[CUBEB_LOG_MESSAGE_MAX_SIZE])
+  {
+    cubeb_log_message msg(str);
+    auto * owned_queue = msg_queue.load();
+    // Check if the queue is being deallocated. If not, grab ownership. If yes,
+    // return, the message won't be logged.
+    if (!owned_queue ||
+        !msg_queue.compare_exchange_strong(owned_queue, nullptr)) {
+      return;
+    }
+    owned_queue->enqueue(msg);
+    // Return ownership.
+    msg_queue.store(owned_queue);
+  }
+  void run()
+  {
+    assert(logging_thread.get_id() == std::thread::id());
+    logging_thread = std::thread([this]() {
+      CUBEB_REGISTER_THREAD("cubeb_log");
+      while (!shutdown_thread) {
+        cubeb_log_message msg;
+        while (msg_queue_consumer.load()->dequeue(&msg, 1)) {
+          cubeb_log_internal_no_format(msg.get());
+        }
+        std::this_thread::sleep_for(
+            std::chrono::milliseconds(CUBEB_LOG_BATCH_PRINT_INTERVAL_MS));
+      }
+      CUBEB_UNREGISTER_THREAD();
+    });
+  }
+  // Tell the underlying queue the producer thread has changed, so it does not
+  // assert in debug. This should be called with the thread stopped.
+  void reset_producer_thread()
+  {
+    if (msg_queue) {
+      msg_queue.load()->reset_thread_ids();
+    }
+  }
+  void start()
+  {
+    auto * queue =
+        new lock_free_queue<cubeb_log_message>(CUBEB_LOG_MESSAGE_QUEUE_DEPTH);
+    msg_queue.store(queue);
+    msg_queue_consumer.store(queue);
+    shutdown_thread = false;
+    run();
+  }
+  void stop()
+  {
+    assert(((g_cubeb_log_callback == cubeb_noop_log_callback) ||
+            !g_cubeb_log_callback) &&
+           "Only call stop after logging has been disabled.");
+    shutdown_thread = true;
+    if (logging_thread.get_id() != std::thread::id()) {
+      logging_thread.join();
+      logging_thread = std::thread();
+      auto * owned_queue = msg_queue.load();
+      // Check if the queue is being used. If not, grab ownership. If yes,
+      // try again shortly. At this point, the logging thread has been joined,
+      // so nothing is going to dequeue.
+      // If there is a valid pointer here, then the real-time audio thread that
+      // logs won't attempt to write into the queue, and instead drop the
+      // message.
+      while (!msg_queue.compare_exchange_weak(owned_queue, nullptr)) {
+      }
+      delete owned_queue;
+      msg_queue_consumer.store(nullptr);
+    }
+  }
+
+private:
+  cubeb_async_logger() {}
+  ~cubeb_async_logger()
+  {
+    assert(logging_thread.get_id() == std::thread::id() &&
+           (g_cubeb_log_callback == cubeb_noop_log_callback ||
+            !g_cubeb_log_callback));
+    if (msg_queue.load()) {
+      delete msg_queue.load();
+    }
+  }
+  /** This is quite a big data structure, but is only instantiated if the
+   * asynchronous logger is used. The two pointers point to the same object, but
+   * the first one can be temporarily null when a message is being enqueued. */
+  std::atomic<lock_free_queue<cubeb_log_message> *> msg_queue = {nullptr};
+
+  std::atomic<lock_free_queue<cubeb_log_message> *> msg_queue_consumer = {
+      nullptr};
+  std::atomic<bool> shutdown_thread = {false};
+  std::thread logging_thread;
+};
+
 void
 cubeb_log_internal(char const * file, uint32_t line, char const * fmt, ...)
 {
@ -49,6 +176,29 @@ cubeb_log_internal_no_format(const char * msg)
  g_cubeb_log_callback.load()(msg);
 }

+void
+cubeb_async_log(char const * fmt, ...)
+{
+  // This is going to copy a 256 bytes array around, which is fine.
+  // We don't want to allocate memory here, because this is made to
+  // be called from a real-time callback.
+  va_list args;
+  va_start(args, fmt);
+  char msg[CUBEB_LOG_MESSAGE_MAX_SIZE];
+  vsnprintf(msg, CUBEB_LOG_MESSAGE_MAX_SIZE, fmt, args);
+  cubeb_async_logger::get().push(msg);
+  va_end(args);
+}
+
+void
+cubeb_async_log_reset_threads(void)
+{
+  if (!g_cubeb_log_callback) {
+    return;
+  }
+  cubeb_async_logger::get().reset_producer_thread();
+}
+
 void
 cubeb_log_set(cubeb_log_level log_level, cubeb_log_callback log_callback)
 {
@ -57,8 +207,15 @@ cubeb_log_set(cubeb_log_level log_level, cubeb_log_callback log_callback)
  // nullptr, to prevent a TOCTOU race between checking the pointer
  if (log_callback && log_level != CUBEB_LOG_DISABLED) {
    g_cubeb_log_callback = log_callback;
+    if (log_level == CUBEB_LOG_VERBOSE) {
+      cubeb_async_logger::get().start();
+    }
  } else if (!log_callback || CUBEB_LOG_DISABLED) {
    g_cubeb_log_callback = cubeb_noop_log_callback;
+    // This returns once the thread has joined.
+    // This is safe even if CUBEB_LOG_VERBOSE was not set; the thread will
+    // simply not be joinable.
+    cubeb_async_logger::get().stop();
  } else {
    assert(false && "Incorrect parameters passed to cubeb_log_set");
  }
--- a/3rdparty/cubeb/src/cubeb_log.h
+++ b/3rdparty/cubeb/src/cubeb_log.h
@ -39,7 +39,12 @@ cubeb_log_get_callback(void);
 void
 cubeb_log_internal_no_format(const char * msg);
 void
-cubeb_log_internal(const char * filename, uint32_t line, const char * fmt, ...);
+cubeb_log_internal(const char * filename, uint32_t line, const char * fmt, ...)
+    PRINTF_FORMAT(3, 4);
+void
+cubeb_async_log(const char * fmt, ...) PRINTF_FORMAT(1, 2);
+void
+cubeb_async_log_reset_threads(void);

 #ifdef __cplusplus
 }
@ -55,9 +60,16 @@ cubeb_log_internal(const char * filename, uint32_t line, const char * fmt, ...);
    }                                                                          \
  } while (0)

+#define ALOG_INTERNAL(level, fmt, ...)                                         \
+  do {                                                                         \
+    if (cubeb_log_get_level() >= level && cubeb_log_get_callback()) {          \
+      cubeb_async_log(fmt, ##__VA_ARGS__);                                     \
+    }                                                                          \
+  } while (0)
+
 /* Asynchronous logging macros to log in real-time callbacks. */
 /* Should not be used on android due to the use of global/static variables. */
-#define ALOGV(msg, ...) LOG_INTERNAL(CUBEB_LOG_VERBOSE, msg, ##__VA_ARGS__)
-#define ALOG(msg, ...) LOG_INTERNAL(CUBEB_LOG_NORMAL, msg, ##__VA_ARGS__)
+#define ALOGV(msg, ...) ALOG_INTERNAL(CUBEB_LOG_VERBOSE, msg, ##__VA_ARGS__)
+#define ALOG(msg, ...) ALOG_INTERNAL(CUBEB_LOG_NORMAL, msg, ##__VA_ARGS__)

 #endif // CUBEB_LOG
--- a/3rdparty/cubeb/src/cubeb_resampler.cpp
+++ b/3rdparty/cubeb/src/cubeb_resampler.cpp
@ -371,3 +371,9 @@ cubeb_resampler_latency(cubeb_resampler * resampler)
 {
  return resampler->latency();
 }
+
+cubeb_resampler_stats
+cubeb_resampler_stats_get(cubeb_resampler * resampler)
+{
+  return resampler->stats();
+}
--- a/3rdparty/cubeb/src/cubeb_resampler.h
+++ b/3rdparty/cubeb/src/cubeb_resampler.h
@ -84,6 +84,20 @@ cubeb_resampler_destroy(cubeb_resampler * resampler);
 long
 cubeb_resampler_latency(cubeb_resampler * resampler);

+/**
+ * Test-only introspection API to ensure that there is no buffering
+ * buildup when resampling.
+ */
+typedef struct {
+  size_t input_input_buffer_size;
+  size_t input_output_buffer_size;
+  size_t output_input_buffer_size;
+  size_t output_output_buffer_size;
+} cubeb_resampler_stats;
+
+cubeb_resampler_stats
+cubeb_resampler_stats_get(cubeb_resampler * resampler);
+
 #if defined(__cplusplus)
 }
 #endif
--- a/3rdparty/cubeb/src/cubeb_resampler_internal.h
+++ b/3rdparty/cubeb/src/cubeb_resampler_internal.h
@ -56,6 +56,7 @@ struct cubeb_resampler {
  virtual long fill(void * input_buffer, long * input_frames_count,
                    void * output_buffer, long frames_needed) = 0;
  virtual long latency() = 0;
+  virtual cubeb_resampler_stats stats() = 0;
  virtual ~cubeb_resampler() {}
 };

@ -86,6 +87,16 @@ public:

  virtual long latency() { return 0; }

+  virtual cubeb_resampler_stats stats()
+  {
+    cubeb_resampler_stats stats;
+    stats.input_input_buffer_size = internal_input_buffer.length();
+    stats.input_output_buffer_size = 0;
+    stats.output_input_buffer_size = 0;
+    stats.output_output_buffer_size = 0;
+    return stats;
+  }
+
  void drop_audio_if_needed()
  {
    uint32_t to_keep = min_buffered_audio_frame(sample_rate);
@ -122,6 +133,20 @@ public:
  virtual long fill(void * input_buffer, long * input_frames_count,
                    void * output_buffer, long output_frames_needed);

+  virtual cubeb_resampler_stats stats()
+  {
+    cubeb_resampler_stats stats = {};
+    if (input_processor) {
+      stats.input_input_buffer_size = input_processor->input_buffer_size();
+      stats.input_output_buffer_size = input_processor->output_buffer_size();
+    }
+    if (output_processor) {
+      stats.output_input_buffer_size = output_processor->input_buffer_size();
+      stats.output_output_buffer_size = output_processor->output_buffer_size();
+    }
+    return stats;
+  }
+
  virtual long latency()
  {
    if (input_processor && output_processor) {
@ -280,29 +305,28 @@ public:
  }

  /** Returns the number of frames to pass in the input of the resampler to have
-   * exactly `output_frame_count` resampled frames. This can return a number
-   * slightly bigger than what is strictly necessary, but it guaranteed that the
-   * number of output frames will be exactly equal. */
+   * at least `output_frame_count` resampled frames. */
  uint32_t input_needed_for_output(int32_t output_frame_count) const
  {
    assert(output_frame_count >= 0); // Check overflow
    int32_t unresampled_frames_left =
        samples_to_frames(resampling_in_buffer.length());
-    int32_t resampled_frames_left =
-        samples_to_frames(resampling_out_buffer.length());
-    float input_frames_needed =
-        (output_frame_count - unresampled_frames_left) * resampling_ratio -
-        resampled_frames_left;
-    if (input_frames_needed < 0) {
-      return 0;
-    }
-    return (uint32_t)ceilf(input_frames_needed);
+    float input_frames_needed_frac =
+        static_cast<float>(output_frame_count) * resampling_ratio;
+    // speex_resample()` can be irregular in its consumption of input samples.
+    // Provide one more frame than the number that would be required with
+    // regular consumption, to make the speex resampler behave more regularly,
+    // and so predictably.
+    auto input_frame_needed =
+        1 + static_cast<int32_t>(ceilf(input_frames_needed_frac));
+    input_frame_needed -= std::min(unresampled_frames_left, input_frame_needed);
+    return input_frame_needed;
  }

  /** Returns a pointer to the input buffer, that contains empty space for at
-   * least `frame_count` elements. This is useful so that consumer can directly
-   * write into the input buffer of the resampler. The pointer returned is
-   * adjusted so that leftover data are not overwritten.
+   * least `frame_count` elements. This is useful so that consumer can
+   * directly write into the input buffer of the resampler. The pointer
+   * returned is adjusted so that leftover data are not overwritten.
   */
  T * input_buffer(size_t frame_count)
  {
@ -312,8 +336,8 @@ public:
    return resampling_in_buffer.data() + leftover_samples;
  }

-  /** This method works with `input_buffer`, and allows to inform the processor
-      how much frames have been written in the provided buffer. */
+  /** This method works with `input_buffer`, and allows to inform the
+     processor how much frames have been written in the provided buffer. */
  void written(size_t written_frames)
  {
    resampling_in_buffer.set_length(leftover_samples +
@ -331,6 +355,9 @@ public:
    }
  }

+  size_t input_buffer_size() const { return resampling_in_buffer.length(); }
+  size_t output_buffer_size() const { return resampling_out_buffer.length(); }
+
 private:
  /** Wrapper for the speex resampling functions to have a typed
   * interface. */
@ -359,6 +386,7 @@ private:
            output_frame_count);
    assert(rv == RESAMPLER_ERR_SUCCESS);
  }
+
  /** The state for the speex resampler used internaly. */
  SpeexResamplerState * speex_resampler;
  /** Source rate / target rate. */
@ -371,8 +399,8 @@ private:
  auto_array<T> resampling_out_buffer;
  /** Additional latency inserted into the pipeline for synchronisation. */
  uint32_t additional_latency;
-  /** When `input_buffer` is called, this allows tracking the number of samples
-      that were in the buffer. */
+  /** When `input_buffer` is called, this allows tracking the number of
+     samples that were in the buffer. */
  uint32_t leftover_samples;
 };

@ -417,8 +445,8 @@ public:
    return delay_output_buffer.data();
  }
  /** Get a pointer to the first writable location in the input buffer>
-   * @parameter frames_needed the number of frames the user needs to write into
-   * the buffer.
+   * @parameter frames_needed the number of frames the user needs to write
+   * into the buffer.
   * @returns a pointer to a location in the input buffer where #frames_needed
   * can be writen. */
  T * input_buffer(uint32_t frames_needed)
@ -428,8 +456,8 @@ public:
                               frames_to_samples(frames_needed));
    return delay_input_buffer.data() + leftover_samples;
  }
-  /** This method works with `input_buffer`, and allows to inform the processor
-      how much frames have been written in the provided buffer. */
+  /** This method works with `input_buffer`, and allows to inform the
+     processor how much frames have been written in the provided buffer. */
  void written(size_t frames_written)
  {
    delay_input_buffer.set_length(leftover_samples +
@ -450,8 +478,8 @@ public:

    return to_pop;
  }
-  /** Returns the number of frames one needs to input into the delay line to get
-   * #frames_needed frames back.
+  /** Returns the number of frames one needs to input into the delay line to
+   * get #frames_needed frames back.
   * @parameter frames_needed the number of frames one want to write into the
   * delay_line
   * @returns the number of frames one will get. */
@ -469,19 +497,23 @@ public:

  void drop_audio_if_needed()
  {
-    size_t available = samples_to_frames(delay_input_buffer.length());
+    uint32_t available = samples_to_frames(delay_input_buffer.length());
    uint32_t to_keep = min_buffered_audio_frame(sample_rate);
    if (available > to_keep) {
      ALOGV("Dropping %u frames", available - to_keep);
+
      delay_input_buffer.pop(nullptr, frames_to_samples(available - to_keep));
    }
  }

+  size_t input_buffer_size() const { return delay_input_buffer.length(); }
+  size_t output_buffer_size() const { return delay_output_buffer.length(); }
+
 private:
  /** The length, in frames, of this delay line */
  uint32_t length;
-  /** When `input_buffer` is called, this allows tracking the number of samples
-      that where in the buffer. */
+  /** When `input_buffer` is called, this allows tracking the number of
+     samples that where in the buffer. */
  uint32_t leftover_samples;
  /** The input buffer, where the delay is applied. */
  auto_array<T> delay_input_buffer;
@ -511,8 +543,8 @@ cubeb_resampler_create_internal(cubeb_stream * stream,
         "need at least one valid parameter pointer.");

  /* All the streams we have have a sample rate that matches the target
-     sample rate, use a no-op resampler, that simply forwards the buffers to the
-     callback. */
+     sample rate, use a no-op resampler, that simply forwards the buffers to
+     the callback. */
  if (((input_params && input_params->rate == target_rate) &&
       (output_params && output_params->rate == target_rate)) ||
      (input_params && !output_params && (input_params->rate == target_rate)) ||
--- a/3rdparty/cubeb/src/cubeb_wasapi.cpp
+++ b/3rdparty/cubeb/src/cubeb_wasapi.cpp
@ -4,8 +4,12 @@
 * This program is made available under an ISC-style license.  See the
 * accompanying file LICENSE for details.
 */
+#ifndef _WIN32_WINNT
 #define _WIN32_WINNT 0x0603
+#endif // !_WIN32_WINNT
+#ifndef NOMINMAX
 #define NOMINMAX
+#endif // !NOMINMAX

 #include <algorithm>
 #include <atomic>
@ -37,31 +41,6 @@
 #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
@ -229,11 +208,6 @@ 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>
@ -304,8 +278,8 @@ wasapi_enumerate_devices_internal(cubeb * context, cubeb_device_type type,
 static int
 wasapi_device_collection_destroy(cubeb * ctx,
                                 cubeb_device_collection * collection);
-static char const *
-wstr_to_utf8(wchar_t const * str);
+static std::unique_ptr<char const[]>
+wstr_to_utf8(LPCWSTR str);
 static std::unique_ptr<wchar_t const[]>
 utf8_to_wstr(char const * str);

@ -314,6 +288,15 @@ utf8_to_wstr(char const * str);
 class wasapi_collection_notification_client;
 class monitor_device_notifications;

+typedef enum {
+  /* Clear options */
+  CUBEB_AUDIO_CLIENT2_NONE,
+  /* Use AUDCLNT_STREAMOPTIONS_RAW  */
+  CUBEB_AUDIO_CLIENT2_RAW,
+  /* Use CUBEB_STREAM_PREF_COMMUNICATIONS */
+  CUBEB_AUDIO_CLIENT2_VOICE
+} AudioClient2Option;
+
 struct cubeb {
  cubeb_ops const * ops = &wasapi_ops;
  owned_critical_section lock;
@ -331,13 +314,6 @@ 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;
@ -360,20 +336,33 @@ struct cubeb_stream {
  /* Mixer pameters. We need to convert the input stream to this
     samplerate/channel layout, as WASAPI does not resample nor upmix
     itself. */
-  cubeb_stream_params input_mix_params = {CUBEB_SAMPLE_FLOAT32NE, 0, 0,
+  cubeb_stream_params input_mix_params = {CUBEB_SAMPLE_FLOAT32NE,
+                                          0,
+                                          0,
                                          CUBEB_LAYOUT_UNDEFINED,
-                                          CUBEB_STREAM_PREF_NONE};
-  cubeb_stream_params output_mix_params = {CUBEB_SAMPLE_FLOAT32NE, 0, 0,
+                                          CUBEB_STREAM_PREF_NONE,
+                                          CUBEB_INPUT_PROCESSING_PARAM_NONE};
+  cubeb_stream_params output_mix_params = {CUBEB_SAMPLE_FLOAT32NE,
+                                           0,
+                                           0,
                                           CUBEB_LAYOUT_UNDEFINED,
-                                           CUBEB_STREAM_PREF_NONE};
+                                           CUBEB_STREAM_PREF_NONE,
+                                           CUBEB_INPUT_PROCESSING_PARAM_NONE};
  /* Stream parameters. This is what the client requested,
   * and what will be presented in the callback. */
-  cubeb_stream_params input_stream_params = {CUBEB_SAMPLE_FLOAT32NE, 0, 0,
+  cubeb_stream_params input_stream_params = {CUBEB_SAMPLE_FLOAT32NE,
+                                             0,
+                                             0,
                                             CUBEB_LAYOUT_UNDEFINED,
-                                             CUBEB_STREAM_PREF_NONE};
-  cubeb_stream_params output_stream_params = {CUBEB_SAMPLE_FLOAT32NE, 0, 0,
+                                             CUBEB_STREAM_PREF_NONE,
+                                             CUBEB_INPUT_PROCESSING_PARAM_NONE};
+  cubeb_stream_params output_stream_params = {
+      CUBEB_SAMPLE_FLOAT32NE,
+      0,
+      0,
      CUBEB_LAYOUT_UNDEFINED,
-                                              CUBEB_STREAM_PREF_NONE};
+      CUBEB_STREAM_PREF_NONE,
+      CUBEB_INPUT_PROCESSING_PARAM_NONE};
  /* A MMDevice role for this stream: either communication or console here. */
  ERole role;
  /* True if this stream will transport voice-data. */
@ -662,6 +651,10 @@ public:
                                                   LPCWSTR device_id)
  {
    LOG("collection: Audio device default changed, id = %S.", device_id);
+
+    /* Default device changes count as device collection changes */
+    monitor_notifications.notify(flow);
+
    return S_OK;
  }

@ -772,7 +765,7 @@ public:
                                                   LPCWSTR device_id)
  {
    LOG("endpoint: Audio device default changed flow=%d role=%d "
-        "new_device_id=%ws.",
+        "new_device_id=%S.",
        flow, role, device_id);

    /* we only support a single stream type for now. */
@ -783,11 +776,13 @@ public:
    DWORD last_change_ms = timeGetTime() - last_device_change;
    bool same_device = default_device_id && device_id &&
                       wcscmp(default_device_id.get(), device_id) == 0;
-    LOG("endpoint: Audio device default changed last_change=%u same_device=%d",
+    LOG("endpoint: Audio device default changed last_change=%lu same_device=%d",
        last_change_ms, same_device);
    if (last_change_ms > DEVICE_CHANGE_DEBOUNCE_MS || !same_device) {
      if (device_id) {
-        default_device_id.reset(_wcsdup(device_id));
+        wchar_t * new_device_id = new wchar_t[wcslen(device_id) + 1];
+        wcscpy(new_device_id, device_id);
+        default_device_id.reset(new_device_id);
      } else {
        default_device_id.reset();
      }
@ -863,16 +858,12 @@ intern_device_id(cubeb * ctx, wchar_t const * id)

  auto_lock lock(ctx->lock);

-  char const * tmp = wstr_to_utf8(id);
+  std::unique_ptr<char const[]> tmp = wstr_to_utf8(id);
  if (!tmp) {
    return nullptr;
  }

-  char const * interned = cubeb_strings_intern(ctx->device_ids, tmp);
-
-  free((void *)tmp);
-
-  return interned;
+  return cubeb_strings_intern(ctx->device_ids, tmp.get());
 }

 bool
@ -977,7 +968,7 @@ refill(cubeb_stream * stm, void * input_buffer, long input_frames_count,
      cubeb_resampler_fill(stm->resampler.get(), input_buffer,
                           &input_frames_count, dest, output_frames_needed);
  if (out_frames < 0) {
-    ALOGV("Callback refill error: %d", out_frames);
+    ALOGV("Callback refill error: %ld", out_frames);
    wasapi_state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR);
    return out_frames;
  }
@ -1263,8 +1254,8 @@ refill_callback_duplex(cubeb_stream * stm)
  XASSERT(has_input(stm) && has_output(stm));

  if (stm->input_stream_params.prefs & CUBEB_STREAM_PREF_LOOPBACK) {
-    HRESULT rv = get_input_buffer(stm);
-    if (FAILED(rv)) {
+    rv = get_input_buffer(stm);
+    if (!rv) {
      return rv;
    }
  }
@ -1274,7 +1265,6 @@ refill_callback_duplex(cubeb_stream * stm)

  rv = get_output_buffer(stm, output_buffer, output_frames);
  if (!rv) {
-    hr = stm->render_client->ReleaseBuffer(output_frames, 0);
    return rv;
  }

@ -1291,9 +1281,11 @@ refill_callback_duplex(cubeb_stream * stm)

  stm->total_output_frames += output_frames;

-  ALOGV("in: %zu, out: %zu, missing: %ld, ratio: %f", stm->total_input_frames,
-        stm->total_output_frames,
-        static_cast<long>(stm->total_output_frames) - stm->total_input_frames,
+  ALOGV("in: %llu, out: %llu, missing: %ld, ratio: %f",
+        (unsigned long long)stm->total_input_frames,
+        (unsigned long long)stm->total_output_frames,
+        static_cast<long long>(stm->total_output_frames) -
+            static_cast<long long>(stm->total_input_frames),
        static_cast<float>(stm->total_output_frames) / stm->total_input_frames);

  long got;
@ -1438,8 +1430,7 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)

  /* We could consider using "Pro Audio" here for WebAudio and
     maybe WebRTC. */
-  mmcss_handle =
-      stm->context->set_mm_thread_characteristics(L"Audio", &mmcss_task_index);
+  mmcss_handle = AvSetMmThreadCharacteristicsA("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",
@ -1519,8 +1510,8 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)
      is_playing = stm->refill_callback(stm);
      break;
    case WAIT_OBJECT_0 + 3: { /* input available */
-      HRESULT rv = get_input_buffer(stm);
-      if (FAILED(rv)) {
+      bool rv = get_input_buffer(stm);
+      if (!rv) {
        is_playing = false;
        continue;
      }
@ -1532,8 +1523,11 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)
      break;
    }
    default:
-      LOG("case %lu not handled in render loop.", waitResult);
-      XASSERT(false);
+      LOG("render_loop: waitResult=%lu (lastError=%lu) unhandled, exiting",
+          waitResult, GetLastError());
+      is_playing = false;
+      hr = E_FAIL;
+      continue;
    }
  }

@ -1547,7 +1541,7 @@ static unsigned int __stdcall wasapi_stream_render_loop(LPVOID stream)
  }

  if (mmcss_handle) {
-    stm->context->revert_mm_thread_characteristics(mmcss_handle);
+    AvRevertMmThreadCharacteristics(mmcss_handle);
  }

  if (FAILED(hr)) {
@ -1560,18 +1554,6 @@ 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)
 {
@ -1807,31 +1789,6 @@ 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;
@ -1839,7 +1796,6 @@ wasapi_init(cubeb ** context, char const * context_name)
 }

 namespace {
-enum ShutdownPhase { OnStop, OnDestroy };

 bool
 stop_and_join_render_thread(cubeb_stream * stm)
@ -1855,16 +1811,7 @@ stop_and_join_render_thread(cubeb_stream * stm)
    return false;
  }

-  /* Wait five seconds for the rendering thread to return. It's supposed to
-   * check its event loop very often, five seconds is rather conservative.
-   * Note: 5*1s loop to work around timer sleep issues on pre-Windows 8. */
-  DWORD r;
-  for (int i = 0; i < 5; ++i) {
-    r = WaitForSingleObject(stm->thread, 1000);
-    if (r == WAIT_OBJECT_0) {
-      break;
-    }
-  }
+  DWORD r = WaitForSingleObject(stm->thread, INFINITE);
  if (r != WAIT_OBJECT_0) {
    LOG("stop_and_join_render_thread: WaitForSingleObject on thread failed: "
        "%lx, %lx",
@ -1888,10 +1835,6 @@ wasapi_destroy(cubeb * context)
    }
  }

-  if (context->mmcss_module) {
-    FreeLibrary(context->mmcss_module);
-  }
-
  delete context;
 }

@ -1949,44 +1892,6 @@ 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());
@ -2006,8 +1911,18 @@ wasapi_get_min_latency(cubeb * ctx, cubeb_stream_params params,
  LOG("default device period: %I64d, minimum device period: %I64d",
      default_period, minimum_period);

-  // The minimum_period is only relevant in exclusive streams.
+  /* 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
  *latency_frames = hns_to_frames(params.rate, default_period);
+#endif

  LOG("Minimum latency in frames: %u", *latency_frames);

@ -2044,6 +1959,21 @@ wasapi_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate)
  return CUBEB_OK;
 }

+int
+wasapi_get_supported_input_processing_params(
+    cubeb * ctx, cubeb_input_processing_params * params)
+{
+  // This is not entirely accurate -- windows doesn't document precisely what
+  // AudioCategory_Communications does -- but assume that we can set all or none
+  // of them.
+  *params = static_cast<cubeb_input_processing_params>(
+      CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
+      CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION |
+      CUBEB_INPUT_PROCESSING_PARAM_AUTOMATIC_GAIN_CONTROL |
+      CUBEB_INPUT_PROCESSING_PARAM_VOICE_ISOLATION);
+  return CUBEB_OK;
+}
+
 static void
 waveformatex_update_derived_properties(WAVEFORMATEX * format)
 {
@ -2097,10 +2027,7 @@ 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.
-       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 */
+       the suggestion, since it seems to always be IEEE_FLOAT. */
    LOG("Using WASAPI suggested format: channels: %d", closest->nChannels);
    XASSERT(closest->wFormatTag == WAVE_FORMAT_EXTENSIBLE);
    WAVEFORMATEXTENSIBLE * closest_pcm =
@ -2122,7 +2049,8 @@ handle_channel_layout(cubeb_stream * stm, EDataFlow direction,
 }

 static int
-initialize_iaudioclient2(com_ptr<IAudioClient> & audio_client)
+initialize_iaudioclient2(com_ptr<IAudioClient> & audio_client,
+                         AudioClient2Option option)
 {
  com_ptr<IAudioClient2> audio_client2;
  audio_client->QueryInterface<IAudioClient2>(audio_client2.receive());
@ -2131,10 +2059,14 @@ initialize_iaudioclient2(com_ptr<IAudioClient> & audio_client)
        "AUDCLNT_STREAMOPTIONS_RAW.");
    return CUBEB_OK;
  }
-  AudioClientProperties properties = {0};
+  AudioClientProperties properties = {};
  properties.cbSize = sizeof(AudioClientProperties);
 #ifndef __MINGW32__
+  if (option == CUBEB_AUDIO_CLIENT2_RAW) {
    properties.Options |= AUDCLNT_STREAMOPTIONS_RAW;
+  } else if (option == CUBEB_AUDIO_CLIENT2_VOICE) {
+    properties.eCategory = AudioCategory_Communications;
+  }
 #endif
  HRESULT hr = audio_client2->SetClientProperties(&properties);
  if (FAILED(hr)) {
@ -2144,12 +2076,12 @@ 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,
-                         REFERENCE_TIME latency_hns)
+                         DWORD flags, EDataFlow direction)
 {
  com_ptr<IAudioClient3> audio_client3;
  audio_client->QueryInterface<IAudioClient3>(audio_client3.receive());
@ -2165,22 +2097,24 @@ 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
-  //   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)).
+  // 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.

  HRESULT hr;
  uint32_t default_period = 0, fundamental_period = 0, min_period = 0,
@ -2192,59 +2126,28 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
    LOG("Could not get shared mode engine period: error: %lx", hr);
    return false;
  }
-  uint32_t requested_latency =
-      hns_to_frames(mix_format->nSamplesPerSec, latency_hns);
-#if !ALLOW_AUDIO_CLIENT_3_LATENCY_OVER_DEFAULT
+  uint32_t requested_latency = stm->latency;
  if (requested_latency >= default_period) {
-    LOG("Requested latency %i equal or greater than default latency %i,"
-        " not using IAudioClient3",
+    LOG("Requested latency %i 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;
  }
-  // Likely unnecessary, but won't hurt
-  if (requested_latency > max_period) {
-    requested_latency = max_period;
-  }
+  requested_latency -= (requested_latency - min_period) % fundamental_period;
  if (requested_latency != old_requested_latency) {
    LOG("Requested latency %i was adjusted to %i", old_requested_latency,
        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) {
@ -2256,37 +2159,22 @@ initialize_iaudioclient3(com_ptr<IAudioClient> & audio_client,
  }

  uint32_t current_period = 0;
-  WAVEFORMATEX * current_format_ptr = nullptr;
+  WAVEFORMATEX * current_format = nullptr;
  // We have to pass a valid WAVEFORMATEX** and not nullptr, otherwise
  // GetCurrentSharedModeEnginePeriod will return E_POINTER
-  hr = audio_client3->GetCurrentSharedModeEnginePeriod(&current_format_ptr,
+  hr = audio_client3->GetCurrentSharedModeEnginePeriod(&current_format,
                                                       &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_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);
+  if (current_period >= default_period) {
+    LOG("Current shared mode engine period %i too high, not using IAudioClient",
+        current_period);
    return false;
  }
-#endif

  hr = audio_client3->InitializeSharedAudioStream(flags, current_period,
                                                  mix_format.get(), NULL);
@ -2299,6 +2187,7 @@ 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")

@ -2322,12 +2211,6 @@ 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.
@ -2364,14 +2247,17 @@ setup_wasapi_stream_one_side(cubeb_stream * stm,

    /* Get a client. We will get all other interfaces we need from
     * this pointer. */
-    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) {
+#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
    }
+#endif

    if (FAILED(hr)) {
      LOG("Could not activate the device to get an audio"
@ -2494,21 +2380,41 @@ setup_wasapi_stream_one_side(cubeb_stream * stm,
  }

  if (stream_params->prefs & CUBEB_STREAM_PREF_RAW) {
-    if (initialize_iaudioclient2(audio_client) != CUBEB_OK) {
+    if (initialize_iaudioclient2(audio_client, CUBEB_AUDIO_CLIENT2_RAW) !=
+        CUBEB_OK) {
      LOG("Can't initialize an IAudioClient2, error: %lx", GetLastError());
      // This is not fatal.
    }
+  } else if (direction == eCapture &&
+             (stream_params->prefs & CUBEB_STREAM_PREF_VOICE) &&
+             stream_params->input_params != CUBEB_INPUT_PROCESSING_PARAM_NONE) {
+    if (stream_params->input_params ==
+        (CUBEB_INPUT_PROCESSING_PARAM_ECHO_CANCELLATION |
+         CUBEB_INPUT_PROCESSING_PARAM_NOISE_SUPPRESSION |
+         CUBEB_INPUT_PROCESSING_PARAM_AUTOMATIC_GAIN_CONTROL |
+         CUBEB_INPUT_PROCESSING_PARAM_VOICE_ISOLATION)) {
+      if (initialize_iaudioclient2(audio_client, CUBEB_AUDIO_CLIENT2_VOICE) !=
+          CUBEB_OK) {
+        LOG("Can't initialize an IAudioClient2, error: %lx", GetLastError());
+        // This is not fatal.
+      }
+    } else {
+      LOG("Invalid combination of input processing params %#x",
+          stream_params->input_params);
+      return CUBEB_ERROR;
+    }
  }

-  if (allow_audio_client_3 &&
-      initialize_iaudioclient3(audio_client, stm, mix_format, flags, direction,
-                               latency_hns)) {
+#if 0 // See https://bugzilla.mozilla.org/show_bug.cgi?id=1590902
+  if (initialize_iaudioclient3(audio_client, stm, mix_format, flags, direction)) {
    LOG("Initialized with IAudioClient3");
  } else {
-    hr = audio_client->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, latency_hns,
-                                  0, mix_format.get(), NULL);
+#endif
+  hr = audio_client->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, latency_hns, 0,
+                                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;
@ -2970,6 +2876,7 @@ wasapi_stream_init(cubeb * context, cubeb_stream ** stream,
    }
  }

+  cubeb_async_log_reset_threads();
  stm->thread =
      (HANDLE)_beginthreadex(NULL, 512 * 1024, wasapi_stream_render_loop, stm,
                             STACK_SIZE_PARAM_IS_A_RESERVATION, NULL);
@ -3031,7 +2938,7 @@ wasapi_stream_add_ref(cubeb_stream * stm)
 {
  XASSERT(stm);
  LONG result = InterlockedIncrement(&stm->ref_count);
-  LOGV("Stream ref count incremented = %i (%p)", result, stm);
+  LOGV("Stream ref count incremented = %ld (%p)", result, stm);
  return result;
 }

@ -3041,7 +2948,7 @@ wasapi_stream_release(cubeb_stream * stm)
  XASSERT(stm);

  LONG result = InterlockedDecrement(&stm->ref_count);
-  LOGV("Stream ref count decremented = %i (%p)", result, stm);
+  LOGV("Stream ref count decremented = %ld (%p)", result, stm);
  if (result == 0) {
    LOG("Stream ref count hit zero, destroying (%p)", stm);

@ -3303,7 +3210,7 @@ wasapi_stream_set_volume(cubeb_stream * stm, float volume)
  return CUBEB_OK;
 }

-static char const *
+static std::unique_ptr<char const[]>
 wstr_to_utf8(LPCWSTR str)
 {
  int size = ::WideCharToMultiByte(CP_UTF8, 0, str, -1, nullptr, 0, NULL, NULL);
@ -3311,8 +3218,8 @@ wstr_to_utf8(LPCWSTR str)
    return nullptr;
  }

-  char * ret = static_cast<char *>(malloc(size));
-  ::WideCharToMultiByte(CP_UTF8, 0, str, -1, ret, size, NULL, NULL);
+  std::unique_ptr<char[]> ret(new char[size]);
+  ::WideCharToMultiByte(CP_UTF8, 0, str, -1, ret.get(), size, NULL, NULL);
  return ret;
 }

@ -3440,7 +3347,7 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
  prop_variant namevar;
  hr = propstore->GetValue(PKEY_Device_FriendlyName, &namevar);
  if (SUCCEEDED(hr) && namevar.vt == VT_LPWSTR) {
-    ret.friendly_name = wstr_to_utf8(namevar.pwszVal);
+    ret.friendly_name = wstr_to_utf8(namevar.pwszVal).release();
  }
  if (!ret.friendly_name) {
    // This is not fatal, but a valid string is expected in all cases.
@ -3461,7 +3368,7 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
    prop_variant instancevar;
    hr = ps->GetValue(PKEY_Device_InstanceId, &instancevar);
    if (SUCCEEDED(hr) && instancevar.vt == VT_LPWSTR) {
-      ret.group_id = wstr_to_utf8(instancevar.pwszVal);
+      ret.group_id = wstr_to_utf8(instancevar.pwszVal).release();
    }
  }

@ -3477,7 +3384,8 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
    ret.preferred =
        (cubeb_device_pref)(ret.preferred | CUBEB_DEVICE_PREF_MULTIMEDIA |
                            CUBEB_DEVICE_PREF_NOTIFICATION);
-  } else if (defaults->is_default(flow, eCommunications, device_id.get())) {
+  }
+  if (defaults->is_default(flow, eCommunications, device_id.get())) {
    ret.preferred =
        (cubeb_device_pref)(ret.preferred | CUBEB_DEVICE_PREF_VOICE);
  }
@ -3504,7 +3412,6 @@ 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)) {
@ -3514,7 +3421,8 @@ 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)) {
-      wfx = reinterpret_cast<WAVEFORMATEX *>(fmtvar.blob.pBlobData);
+      WAVEFORMATEX * wfx =
+          reinterpret_cast<WAVEFORMATEX *>(fmtvar.blob.pBlobData);

      if (fmtvar.blob.cbSize >= sizeof(WAVEFORMATEX) + wfx->cbSize ||
          wfx->wFormatTag == WAVE_FORMAT_PCM) {
@ -3524,27 +3432,6 @@ wasapi_create_device(cubeb * ctx, cubeb_device_info & ret,
    }
  }

-#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))) {
@ -3638,7 +3525,7 @@ wasapi_enumerate_devices(cubeb * context, cubeb_device_type type,
 {
  return wasapi_enumerate_devices_internal(
      context, type, out,
-      DEVICE_STATE_ACTIVE /*| DEVICE_STATE_DISABLED | DEVICE_STATE_UNPLUGGED*/);
+      DEVICE_STATE_ACTIVE | DEVICE_STATE_DISABLED | DEVICE_STATE_UNPLUGGED);
 }

 static int
@ -3656,6 +3543,14 @@ wasapi_device_collection_destroy(cubeb * /*ctx*/,
  return CUBEB_OK;
 }

+int
+wasapi_set_input_processing_params(cubeb_stream * stream,
+                                   cubeb_input_processing_params params)
+{
+  LOG("Cannot set voice processing params after init. Use cubeb_stream_init.");
+  return CUBEB_ERROR_NOT_SUPPORTED;
+}
+
 static int
 wasapi_register_device_collection_changed(
    cubeb * context, cubeb_device_type devtype,
@ -3736,7 +3631,8 @@ 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,
+    /*.get_supported_input_processing_params =*/
+    wasapi_get_supported_input_processing_params,
    /*.enumerate_devices =*/wasapi_enumerate_devices,
    /*.device_collection_destroy =*/wasapi_device_collection_destroy,
    /*.destroy =*/wasapi_destroy,
@ -3751,7 +3647,7 @@ cubeb_ops const wasapi_ops = {
    /*.stream_set_name =*/NULL,
    /*.stream_get_current_device =*/NULL,
    /*.stream_set_input_mute =*/NULL,
-    /*.stream_set_input_processing_params =*/NULL,
+    /*.stream_set_input_processing_params =*/wasapi_set_input_processing_params,
    /*.stream_device_destroy =*/NULL,
    /*.stream_register_device_changed_callback =*/NULL,
    /*.register_device_collection_changed =*/
--- a/3rdparty/cubeb/subprojects/speex/arch.h
+++ b/3rdparty/cubeb/subprojects/speex/arch.h
@ -41,10 +41,10 @@
 #ifdef FLOATING_POINT
 #error You cannot compile as floating point and fixed point at the same time
 #endif
-#ifdef _USE_SSE
+#ifdef USE_SSE
 #error SSE is only for floating-point
 #endif
-#if ((defined (ARM4_ASM)||defined (ARM4_ASM)) && defined(BFIN_ASM)) || (defined (ARM4_ASM)&&defined(ARM5E_ASM))
+#if defined(ARM4_ASM) + defined(ARM5E_ASM) + defined(BFIN_ASM) > 1
 #error Make up your mind. What CPU do you have?
 #endif
 #ifdef VORBIS_PSYCHO
@ -56,10 +56,10 @@
 #ifndef FLOATING_POINT
 #error You now need to define either FIXED_POINT or FLOATING_POINT
 #endif
-#if defined (ARM4_ASM) || defined(ARM5E_ASM) || defined(BFIN_ASM)
+#if defined(ARM4_ASM) || defined(ARM5E_ASM) || defined(BFIN_ASM)
 #error I suppose you can have a [ARM4/ARM5E/Blackfin] that has float instructions?
 #endif
-#ifdef FIXED_POINT_DEBUG
+#ifdef FIXED_DEBUG
 #error "Don't you think enabling fixed-point is a good thing to do if you want to debug that?"
 #endif

@ -117,9 +117,9 @@ typedef spx_word32_t spx_sig_t;

 #ifdef ARM5E_ASM
 #include "fixed_arm5e.h"
-#elif defined (ARM4_ASM)
+#elif defined(ARM4_ASM)
 #include "fixed_arm4.h"
-#elif defined (BFIN_ASM)
+#elif defined(BFIN_ASM)
 #include "fixed_bfin.h"
 #endif

@ -177,16 +177,13 @@ typedef float spx_word32_t;
 #define ADD32(a,b) ((a)+(b))
 #define SUB32(a,b) ((a)-(b))
 #define MULT16_16_16(a,b)     ((a)*(b))
+#define MULT16_32_32(a,b)     ((a)*(b))
 #define MULT16_16(a,b)     ((spx_word32_t)(a)*(spx_word32_t)(b))
 #define MAC16_16(c,a,b)     ((c)+(spx_word32_t)(a)*(spx_word32_t)(b))

-#define MULT16_32_Q11(a,b)     ((a)*(b))
-#define MULT16_32_Q13(a,b)     ((a)*(b))
-#define MULT16_32_Q14(a,b)     ((a)*(b))
 #define MULT16_32_Q15(a,b)     ((a)*(b))
 #define MULT16_32_P15(a,b)     ((a)*(b))

-#define MAC16_32_Q11(c,a,b)     ((c)+(a)*(b))
 #define MAC16_32_Q15(c,a,b)     ((c)+(a)*(b))

 #define MAC16_16_Q11(c,a,b)     ((c)+(a)*(b))
@ -210,7 +207,7 @@ typedef float spx_word32_t;
 #endif


-#if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+#if defined(CONFIG_TI_C54X) || defined(CONFIG_TI_C55X)

 /* 2 on TI C5x DSP */
 #define BYTES_PER_CHAR 2
--- a/3rdparty/cubeb/subprojects/speex/fixed_generic.h
+++ b/3rdparty/cubeb/subprojects/speex/fixed_generic.h
@ -69,22 +69,18 @@


 /* result fits in 16 bits */
-#define MULT16_16_16(a,b)     ((((spx_word16_t)(a))*((spx_word16_t)(b))))
+#define MULT16_16_16(a,b)     (((spx_word16_t)(a))*((spx_word16_t)(b)))
+/* result fits in 32 bits */
+#define MULT16_32_32(a,b)     (((spx_word16_t)(a))*((spx_word32_t)(b)))

 /* (spx_word32_t)(spx_word16_t) gives TI compiler a hint that it's 16x16->32 multiply */
 #define MULT16_16(a,b)     (((spx_word32_t)(spx_word16_t)(a))*((spx_word32_t)(spx_word16_t)(b)))

 #define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b))))
-#define MULT16_32_Q12(a,b) ADD32(MULT16_16((a),SHR((b),12)), SHR(MULT16_16((a),((b)&0x00000fff)),12))
-#define MULT16_32_Q13(a,b) ADD32(MULT16_16((a),SHR((b),13)), SHR(MULT16_16((a),((b)&0x00001fff)),13))
-#define MULT16_32_Q14(a,b) ADD32(MULT16_16((a),SHR((b),14)), SHR(MULT16_16((a),((b)&0x00003fff)),14))

-#define MULT16_32_Q11(a,b) ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11))
-#define MAC16_32_Q11(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11)))
-
-#define MULT16_32_P15(a,b) ADD32(MULT16_16((a),SHR((b),15)), PSHR(MULT16_16((a),((b)&0x00007fff)),15))
-#define MULT16_32_Q15(a,b) ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))
-#define MAC16_32_Q15(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15)))
+#define MULT16_32_P15(a,b) ADD32(MULT16_32_32(a,SHR((b),15)), PSHR(MULT16_16((a),((b)&0x00007fff)),15))
+#define MULT16_32_Q15(a,b) ADD32(MULT16_32_32(a,SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))
+#define MAC16_32_Q15(c,a,b) ADD32(c,MULT16_32_Q15(a,b))


 #define MAC16_16_Q11(c,a,b)     (ADD32((c),SHR(MULT16_16((a),(b)),11)))
--- a/3rdparty/cubeb/subprojects/speex/resample.c
+++ b/3rdparty/cubeb/subprojects/speex/resample.c
@ -46,7 +46,7 @@
   Smith, Julius O. Digital Audio Resampling Home Page
   Center for Computer Research in Music and Acoustics (CCRMA),
   Stanford University, 2007.
-   Web published at http://ccrma.stanford.edu/~jos/resample/.
+   Web published at https://ccrma.stanford.edu/~jos/resample/.

   There is one main difference, though. This resampler uses cubic
   interpolation instead of linear interpolation in the above paper. This
@ -63,9 +63,12 @@

 #ifdef OUTSIDE_SPEEX
 #include <stdlib.h>
-static void *speex_alloc (int size) {return calloc(size,1);}
-static void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);}
-static void speex_free (void *ptr) {free(ptr);}
+static void *speex_alloc(int size) {return calloc(size,1);}
+static void *speex_realloc(void *ptr, int size) {return realloc(ptr, size);}
+static void speex_free(void *ptr) {free(ptr);}
+#ifndef EXPORT
+#define EXPORT
+#endif
 #include "speex_resampler.h"
 #include "arch.h"
 #else /* OUTSIDE_SPEEX */
@ -75,7 +78,6 @@ static void speex_free (void *ptr) {free(ptr);}
 #include "os_support.h"
 #endif /* OUTSIDE_SPEEX */

-#include "stack_alloc.h"
 #include <math.h>
 #include <limits.h>

@ -91,18 +93,18 @@ static void speex_free (void *ptr) {free(ptr);}
 #endif

 #ifndef UINT32_MAX
-#define UINT32_MAX 4294967296U
+#define UINT32_MAX 4294967295U
 #endif

-#ifdef _USE_SSE
+#ifdef USE_SSE
 #include "resample_sse.h"
 #endif

-#ifdef _USE_NEON
+#ifdef USE_NEON
 #include "resample_neon.h"
 #endif

-/* Numer of elements to allocate on the stack */
+/* Number of elements to allocate on the stack */
 #ifdef VAR_ARRAYS
 #define FIXED_STACK_ALLOC 8192
 #else
@ -194,16 +196,14 @@ struct FuncDef {
   int oversample;
 };

-static const struct FuncDef _KAISER12 = {kaiser12_table, 64};
-#define KAISER12 (&_KAISER12)
-/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
-#define KAISER12 (&_KAISER12)*/
-static const struct FuncDef _KAISER10 = {kaiser10_table, 32};
-#define KAISER10 (&_KAISER10)
-static const struct FuncDef _KAISER8 = {kaiser8_table, 32};
-#define KAISER8 (&_KAISER8)
-static const struct FuncDef _KAISER6 = {kaiser6_table, 32};
-#define KAISER6 (&_KAISER6)
+static const struct FuncDef kaiser12_funcdef = {kaiser12_table, 64};
+#define KAISER12 (&kaiser12_funcdef)
+static const struct FuncDef kaiser10_funcdef = {kaiser10_table, 32};
+#define KAISER10 (&kaiser10_funcdef)
+static const struct FuncDef kaiser8_funcdef = {kaiser8_table, 32};
+#define KAISER8 (&kaiser8_funcdef)
+static const struct FuncDef kaiser6_funcdef = {kaiser6_table, 32};
+#define KAISER6 (&kaiser6_funcdef)

 struct QualityMapping {
   int base_length;
@ -473,7 +473,7 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint3
      }

      cubic_coef(frac, interp);
-      sum = MULT16_32_Q15(interp[0],SHR32(accum[0], 1)) + MULT16_32_Q15(interp[1],SHR32(accum[1], 1)) + MULT16_32_Q15(interp[2],SHR32(accum[2], 1)) + MULT16_32_Q15(interp[3],SHR32(accum[3], 1));
+      sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);
      sum = SATURATE32PSHR(sum, 15, 32767);
 #else
      cubic_coef(frac, interp);
@ -572,6 +572,7 @@ static int resampler_basic_zero(SpeexResamplerState *st, spx_uint32_t channel_in
   const int frac_advance = st->frac_advance;
   const spx_uint32_t den_rate = st->den_rate;

+   (void)in;
   while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
   {
      out[out_stride * out_sample++] = 0;
@ -589,16 +590,15 @@ static int resampler_basic_zero(SpeexResamplerState *st, spx_uint32_t channel_in
   return out_sample;
 }

-static int _muldiv(spx_uint32_t *result, spx_uint32_t value, spx_uint32_t mul, spx_uint32_t div)
+static int multiply_frac(spx_uint32_t *result, spx_uint32_t value, spx_uint32_t num, spx_uint32_t den)
 {
-   speex_assert(result);
-   spx_uint32_t major = value / div;
-   spx_uint32_t remainder = value % div;
+   spx_uint32_t major = value / den;
+   spx_uint32_t remain = value % den;
   /* TODO: Could use 64 bits operation to check for overflow. But only guaranteed in C99+ */
-   if (remainder > UINT32_MAX / mul || major > UINT32_MAX / mul
-       || major * mul > UINT32_MAX - remainder * mul / div)
+   if (remain > UINT32_MAX / num || major > UINT32_MAX / num
+       || major * num > UINT32_MAX - remain * num / den)
      return RESAMPLER_ERR_OVERFLOW;
-   *result = remainder * mul / div + major * mul;
+   *result = remain * num / den + major * num;
   return RESAMPLER_ERR_SUCCESS;
 }

@ -619,7 +619,7 @@ static int update_filter(SpeexResamplerState *st)
   {
      /* down-sampling */
      st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate;
-      if (_muldiv(&st->filt_len,st->filt_len,st->num_rate,st->den_rate) != RESAMPLER_ERR_SUCCESS)
+      if (multiply_frac(&st->filt_len,st->filt_len,st->num_rate,st->den_rate) != RESAMPLER_ERR_SUCCESS)
         goto fail;
      /* Round up to make sure we have a multiple of 8 for SSE */
      st->filt_len = ((st->filt_len-1)&(~0x7))+8;
@ -638,12 +638,12 @@ static int update_filter(SpeexResamplerState *st)
      st->cutoff = quality_map[st->quality].upsample_bandwidth;
   }

-   /* Choose the resampling type that requires the least amount of memory */
 #ifdef RESAMPLE_FULL_SINC_TABLE
   use_direct = 1;
   if (INT_MAX/sizeof(spx_word16_t)/st->den_rate < st->filt_len)
      goto fail;
 #else
+   /* Choose the resampling type that requires the least amount of memory */
   use_direct = st->filt_len*st->den_rate <= st->filt_len*st->oversample+8
                && INT_MAX/sizeof(spx_word16_t)/st->den_rate >= st->filt_len;
 #endif
@ -733,16 +733,18 @@ static int update_filter(SpeexResamplerState *st)
      {
         spx_uint32_t j;
         spx_uint32_t olen = old_length;
+         spx_uint32_t start = i*st->mem_alloc_size;
+         spx_uint32_t magic_samples = st->magic_samples[i];
         /*if (st->magic_samples[i])*/
         {
            /* Try and remove the magic samples as if nothing had happened */

            /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
-            olen = old_length + 2*st->magic_samples[i];
-            for (j=old_length-1+st->magic_samples[i];j--;)
-               st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]] = st->mem[i*old_alloc_size+j];
-            for (j=0;j<st->magic_samples[i];j++)
-               st->mem[i*st->mem_alloc_size+j] = 0;
+            olen = old_length + 2*magic_samples;
+            for (j=old_length-1+magic_samples;j--;)
+               st->mem[start+j+magic_samples] = st->mem[i*old_alloc_size+j];
+            for (j=0;j<magic_samples;j++)
+               st->mem[start+j] = 0;
            st->magic_samples[i] = 0;
         }
         if (st->filt_len > olen)
@ -750,17 +752,18 @@ static int update_filter(SpeexResamplerState *st)
            /* If the new filter length is still bigger than the "augmented" length */
            /* Copy data going backward */
            for (j=0;j<olen-1;j++)
-               st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*st->mem_alloc_size+(olen-2-j)];
+               st->mem[start+(st->filt_len-2-j)] = st->mem[start+(olen-2-j)];
            /* Then put zeros for lack of anything better */
            for (;j<st->filt_len-1;j++)
-               st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = 0;
+               st->mem[start+(st->filt_len-2-j)] = 0;
            /* Adjust last_sample */
            st->last_sample[i] += (st->filt_len - olen)/2;
         } else {
            /* Put back some of the magic! */
-            st->magic_samples[i] = (olen - st->filt_len)/2;
-            for (j=0;j<st->filt_len-1+st->magic_samples[i];j++)
-               st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
+            magic_samples = (olen - st->filt_len)/2;
+            for (j=0;j<st->filt_len-1+magic_samples;j++)
+               st->mem[start+j] = st->mem[start+j+magic_samples];
+            st->magic_samples[i] = magic_samples;
         }
      }
   } else if (st->filt_len < old_length)
@ -977,8 +980,7 @@ EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t cha
   const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1);
 #ifdef VAR_ARRAYS
   const unsigned int ylen = (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC;
-   VARDECL(spx_word16_t *ystack);
-   ALLOC(ystack, ylen, spx_word16_t);
+   spx_word16_t ystack[ylen];
 #else
   const unsigned int ylen = FIXED_STACK_ALLOC;
   spx_word16_t ystack[FIXED_STACK_ALLOC];
@ -1093,7 +1095,7 @@ EXPORT void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_r
   *out_rate = st->out_rate;
 }

-static inline spx_uint32_t _gcd(spx_uint32_t a, spx_uint32_t b)
+static inline spx_uint32_t compute_gcd(spx_uint32_t a, spx_uint32_t b)
 {
   while (b != 0)
   {
@ -1123,7 +1125,7 @@ EXPORT int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t r
   st->num_rate = ratio_num;
   st->den_rate = ratio_den;

-   fact = _gcd (st->num_rate, st->den_rate);
+   fact = compute_gcd(st->num_rate, st->den_rate);

   st->num_rate /= fact;
   st->den_rate /= fact;
@ -1132,7 +1134,7 @@ EXPORT int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t r
   {
      for (i=0;i<st->nb_channels;i++)
      {
-         if (_muldiv(&st->samp_frac_num[i],st->samp_frac_num[i],st->den_rate,old_den) != RESAMPLER_ERR_SUCCESS)
+         if (multiply_frac(&st->samp_frac_num[i],st->samp_frac_num[i],st->den_rate,old_den) != RESAMPLER_ERR_SUCCESS)
            return RESAMPLER_ERR_OVERFLOW;
         /* Safety net */
         if (st->samp_frac_num[i] >= st->den_rate)
--- a/3rdparty/cubeb/subprojects/speex/resample_neon.h
+++ b/3rdparty/cubeb/subprojects/speex/resample_neon.h
@ -36,14 +36,26 @@
   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 */

-#include <arm_neon.h>
+#include <stdint.h>

 #ifdef FIXED_POINT
-#ifdef __thumb2__
+#if defined(__aarch64__)
+static inline int32_t saturate_32bit_to_16bit(int32_t a) {
+    int32_t ret;
+    asm ("fmov s0, %w[a]\n"
+         "sqxtn h0, s0\n"
+         "sxtl v0.4s, v0.4h\n"
+         "fmov %w[ret], s0\n"
+         : [ret] "=r" (ret)
+         : [a] "r" (a)
+         : "v0" );
+    return ret;
+}
+#elif defined(__thumb2__)
 static inline int32_t saturate_32bit_to_16bit(int32_t a) {
    int32_t ret;
    asm ("ssat %[ret], #16, %[a]"
-         : [ret] "=&r" (ret)
+         : [ret] "=r" (ret)
         : [a] "r" (a)
         : );
    return ret;
@ -54,7 +66,7 @@ static inline int32_t saturate_32bit_to_16bit(int32_t a) {
    asm ("vmov.s32 d0[0], %[a]\n"
         "vqmovn.s32 d0, q0\n"
         "vmov.s16 %[ret], d0[0]\n"
-         : [ret] "=&r" (ret)
+         : [ret] "=r" (ret)
         : [a] "r" (a)
         : "q0");
    return ret;
@ -64,7 +76,63 @@ static inline int32_t saturate_32bit_to_16bit(int32_t a) {
 #define WORD2INT(x) (saturate_32bit_to_16bit(x))

 #define OVERRIDE_INNER_PRODUCT_SINGLE
-/* Only works when len % 4 == 0 */
+/* Only works when len % 4 == 0 and len >= 4 */
+#if defined(__aarch64__)
+static inline int32_t inner_product_single(const int16_t *a, const int16_t *b, unsigned int len)
+{
+    int32_t ret;
+    uint32_t remainder = len % 16;
+    len = len - remainder;
+
+    asm volatile ("	 cmp %w[len], #0\n"
+		  "	 b.ne 1f\n"
+		  "	 ld1 {v16.4h}, [%[b]], #8\n"
+		  "	 ld1 {v20.4h}, [%[a]], #8\n"
+		  "	 subs %w[remainder], %w[remainder], #4\n"
+		  "	 smull v0.4s, v16.4h, v20.4h\n"
+		  "      b.ne 4f\n"
+		  "	 b 5f\n"
+		  "1:"
+		  "	 ld1 {v16.4h, v17.4h, v18.4h, v19.4h}, [%[b]], #32\n"
+		  "	 ld1 {v20.4h, v21.4h, v22.4h, v23.4h}, [%[a]], #32\n"
+		  "	 subs %w[len], %w[len], #16\n"
+		  "	 smull v0.4s, v16.4h, v20.4h\n"
+		  "	 smlal v0.4s, v17.4h, v21.4h\n"
+		  "	 smlal v0.4s, v18.4h, v22.4h\n"
+		  "	 smlal v0.4s, v19.4h, v23.4h\n"
+		  "	 b.eq 3f\n"
+		  "2:"
+		  "	 ld1 {v16.4h, v17.4h, v18.4h, v19.4h}, [%[b]], #32\n"
+		  "	 ld1 {v20.4h, v21.4h, v22.4h, v23.4h}, [%[a]], #32\n"
+		  "	 subs %w[len], %w[len], #16\n"
+		  "	 smlal v0.4s, v16.4h, v20.4h\n"
+		  "	 smlal v0.4s, v17.4h, v21.4h\n"
+		  "	 smlal v0.4s, v18.4h, v22.4h\n"
+		  "	 smlal v0.4s, v19.4h, v23.4h\n"
+		  "	 b.ne 2b\n"
+		  "3:"
+		  "	 cmp %w[remainder], #0\n"
+		  "	 b.eq 5f\n"
+		  "4:"
+		  "	 ld1 {v18.4h}, [%[b]], #8\n"
+		  "	 ld1 {v22.4h}, [%[a]], #8\n"
+		  "	 subs %w[remainder], %w[remainder], #4\n"
+		  "	 smlal v0.4s, v18.4h, v22.4h\n"
+		  "	 b.ne 4b\n"
+		  "5:"
+		  "	 saddlv d0, v0.4s\n"
+		  "	 sqxtn s0, d0\n"
+		  "	 sqrshrn h0, s0, #15\n"
+		  "	 sxtl v0.4s, v0.4h\n"
+		  "	 fmov %w[ret], s0\n"
+		  : [ret] "=r" (ret), [a] "+r" (a), [b] "+r" (b),
+		    [len] "+r" (len), [remainder] "+r" (remainder)
+		  :
+		  : "cc", "v0",
+		    "v16", "v17", "v18", "v19", "v20", "v21", "v22", "v23");
+    return ret;
+}
+#else
 static inline int32_t inner_product_single(const int16_t *a, const int16_t *b, unsigned int len)
 {
    int32_t ret;
@ -112,33 +180,104 @@ static inline int32_t inner_product_single(const int16_t *a, const int16_t *b, u
 		  "	 vqmovn.s64 d0, q0\n"
 		  "	 vqrshrn.s32 d0, q0, #15\n"
 		  "	 vmov.s16 %[ret], d0[0]\n"
-		  : [ret] "=&r" (ret), [a] "+r" (a), [b] "+r" (b),
+		  : [ret] "=r" (ret), [a] "+r" (a), [b] "+r" (b),
 		    [len] "+r" (len), [remainder] "+r" (remainder)
 		  :
 		  : "cc", "q0",
-		    "d16", "d17", "d18", "d19",
-		    "d20", "d21", "d22", "d23");
+		    "d16", "d17", "d18", "d19", "d20", "d21", "d22", "d23");

    return ret;
 }
-#elif defined(FLOATING_POINT)
+#endif  // !defined(__aarch64__)

+#elif defined(FLOATING_POINT)
+#if defined(__aarch64__)
+static inline int32_t saturate_float_to_16bit(float a) {
+    int32_t ret;
+    asm ("fcvtas s1, %s[a]\n"
+         "sqxtn h1, s1\n"
+         "sxtl v1.4s, v1.4h\n"
+         "fmov %w[ret], s1\n"
+         : [ret] "=r" (ret)
+         : [a] "w" (a)
+         : "v1");
+    return ret;
+}
+#else
 static inline int32_t saturate_float_to_16bit(float a) {
    int32_t ret;
    asm ("vmov.f32 d0[0], %[a]\n"
         "vcvt.s32.f32 d0, d0, #15\n"
         "vqrshrn.s32 d0, q0, #15\n"
         "vmov.s16 %[ret], d0[0]\n"
-         : [ret] "=&r" (ret)
+         : [ret] "=r" (ret)
         : [a] "r" (a)
         : "q0");
    return ret;
 }
+#endif
+
 #undef WORD2INT
 #define WORD2INT(x) (saturate_float_to_16bit(x))

 #define OVERRIDE_INNER_PRODUCT_SINGLE
-/* Only works when len % 4 == 0 */
+/* Only works when len % 4 == 0 and len >= 4 */
+#if defined(__aarch64__)
+static inline float inner_product_single(const float *a, const float *b, unsigned int len)
+{
+    float ret;
+    uint32_t remainder = len % 16;
+    len = len - remainder;
+
+    asm volatile ("	 cmp %w[len], #0\n"
+		  "	 b.ne 1f\n"
+		  "	 ld1 {v16.4s}, [%[b]], #16\n"
+		  "	 ld1 {v20.4s}, [%[a]], #16\n"
+		  "	 subs %w[remainder], %w[remainder], #4\n"
+		  "	 fmul v1.4s, v16.4s, v20.4s\n"
+		  "      b.ne 4f\n"
+		  "	 b 5f\n"
+		  "1:"
+		  "	 ld1 {v16.4s, v17.4s, v18.4s, v19.4s}, [%[b]], #64\n"
+		  "	 ld1 {v20.4s, v21.4s, v22.4s, v23.4s}, [%[a]], #64\n"
+		  "	 subs %w[len], %w[len], #16\n"
+		  "	 fmul v1.4s, v16.4s, v20.4s\n"
+		  "	 fmul v2.4s, v17.4s, v21.4s\n"
+		  "	 fmul v3.4s, v18.4s, v22.4s\n"
+		  "	 fmul v4.4s, v19.4s, v23.4s\n"
+		  "	 b.eq 3f\n"
+		  "2:"
+		  "	 ld1 {v16.4s, v17.4s, v18.4s, v19.4s}, [%[b]], #64\n"
+		  "	 ld1 {v20.4s, v21.4s, v22.4s, v23.4s}, [%[a]], #64\n"
+		  "	 subs %w[len], %w[len], #16\n"
+		  "	 fmla v1.4s, v16.4s, v20.4s\n"
+		  "	 fmla v2.4s, v17.4s, v21.4s\n"
+		  "	 fmla v3.4s, v18.4s, v22.4s\n"
+		  "	 fmla v4.4s, v19.4s, v23.4s\n"
+		  "	 b.ne 2b\n"
+		  "3:"
+		  "	 fadd v16.4s, v1.4s, v2.4s\n"
+		  "	 fadd v17.4s, v3.4s, v4.4s\n"
+		  "	 cmp %w[remainder], #0\n"
+		  "	 fadd v1.4s, v16.4s, v17.4s\n"
+		  "	 b.eq 5f\n"
+		  "4:"
+		  "	 ld1 {v18.4s}, [%[b]], #16\n"
+		  "	 ld1 {v22.4s}, [%[a]], #16\n"
+		  "	 subs %w[remainder], %w[remainder], #4\n"
+		  "	 fmla v1.4s, v18.4s, v22.4s\n"
+		  "	 b.ne 4b\n"
+		  "5:"
+		  "	 faddp v1.4s, v1.4s, v1.4s\n"
+		  "	 faddp %[ret].4s, v1.4s, v1.4s\n"
+		  : [ret] "=w" (ret), [a] "+r" (a), [b] "+r" (b),
+		    [len] "+r" (len), [remainder] "+r" (remainder)
+		  :
+		  : "cc", "v1", "v2", "v3", "v4",
+		    "v16", "v17", "v18", "v19", "v20", "v21", "v22", "v23");
+    return ret;
+}
+#else
 static inline float inner_product_single(const float *a, const float *b, unsigned int len)
 {
    float ret;
@ -191,11 +330,12 @@ static inline float inner_product_single(const float *a, const float *b, unsigne
 		  "	 vadd.f32 d0, d0, d1\n"
 		  "	 vpadd.f32 d0, d0, d0\n"
 		  "	 vmov.f32 %[ret], d0[0]\n"
-		  : [ret] "=&r" (ret), [a] "+r" (a), [b] "+r" (b),
+		  : [ret] "=r" (ret), [a] "+r" (a), [b] "+r" (b),
 		    [len] "+l" (len), [remainder] "+l" (remainder)
 		  :
-		  : "cc", "q0", "q1", "q2", "q3", "q4", "q5", "q6", "q7", "q8",
-                    "q9", "q10", "q11");
+		  : "cc", "q0", "q1", "q2", "q3",
+		    "q4", "q5", "q6", "q7", "q8", "q9", "q10", "q11");
    return ret;
 }
+#endif  // defined(__aarch64__)
 #endif
--- a/3rdparty/cubeb/subprojects/speex/resample_sse.h
+++ b/3rdparty/cubeb/subprojects/speex/resample_sse.h
@ -71,7 +71,7 @@ static inline float interpolate_product_single(const float *a, const float *b, u
   return ret;
 }

-#ifdef _USE_SSE2
+#ifdef USE_SSE2
 #include <emmintrin.h>
 #define OVERRIDE_INNER_PRODUCT_DOUBLE

--- a/pcsx2/Host/CubebAudioStream.cpp
+++ b/pcsx2/Host/CubebAudioStream.cpp
@ -288,9 +288,9 @@ std::vector<std::pair<std::string, std::string>> AudioStream::GetCubebDriverName
 	std::vector<std::pair<std::string, std::string>> names;
 	names.emplace_back(std::string(), TRANSLATE_STR("AudioStream", "Default"));

-	const char** cubeb_names = cubeb_get_backend_names();
-	for (u32 i = 0; cubeb_names[i] != nullptr; i++)
-		names.emplace_back(cubeb_names[i], cubeb_names[i]);
+	auto cubeb_names = cubeb_get_backend_names();
+	for (int i = 0; i < cubeb_names.count; i++)
+		names.emplace_back(cubeb_names.names[i], cubeb_names.names[i]);

 	return names;
 }