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Mypal/media/libcubeb/src/cubeb_wasapi.cpp

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/*
* Copyright © 2013 Mozilla Foundation
*
* This program is made available under an ISC-style license. See the
* accompanying file LICENSE for details.
*/
#define NOMINMAX
#include <initguid.h>
#include <windows.h>
#include <mmdeviceapi.h>
#include <windef.h>
#include <audioclient.h>
#include <devicetopology.h>
#include <process.h>
#include <avrt.h>
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <cmath>
#include <algorithm>
#include <memory>
#include <limits>
#include <atomic>
#include "cubeb/cubeb.h"
#include "cubeb-internal.h"
#include "cubeb_resampler.h"
#include "cubeb_utils.h"
/* devicetopology.h missing in MinGW. */
#ifndef __devicetopology_h__
#include "cubeb_devicetopology.h"
#endif
/* Taken from winbase.h, Not in MinGW. */
#ifndef STACK_SIZE_PARAM_IS_A_RESERVATION
#define STACK_SIZE_PARAM_IS_A_RESERVATION 0x00010000 // Threads only
#endif
#ifndef PKEY_Device_FriendlyName
DEFINE_PROPERTYKEY(PKEY_Device_FriendlyName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, 14); // DEVPROP_TYPE_STRING
#endif
#ifndef PKEY_Device_InstanceId
DEFINE_PROPERTYKEY(PKEY_Device_InstanceId, 0x78c34fc8, 0x104a, 0x4aca, 0x9e, 0xa4, 0x52, 0x4d, 0x52, 0x99, 0x6e, 0x57, 0x00000100); // VT_LPWSTR
#endif
namespace {
template<typename T, size_t N>
constexpr size_t
ARRAY_LENGTH(T(&)[N])
{
return N;
}
void
SafeRelease(HANDLE handle)
{
if (handle) {
CloseHandle(handle);
}
}
template <typename T>
void SafeRelease(T * ptr)
{
if (ptr) {
ptr->Release();
}
}
struct auto_com {
auto_com() {
result = CoInitializeEx(NULL, COINIT_MULTITHREADED);
}
~auto_com() {
if (result == RPC_E_CHANGED_MODE) {
// This is not an error, COM was not initialized by this function, so it is
// not necessary to uninit it.
LOG("COM was already initialized in STA.");
} else if (result == S_FALSE) {
// This is not an error. We are allowed to call CoInitializeEx more than
// once, as long as it is matches by an CoUninitialize call.
// We do that in the dtor which is guaranteed to be called.
LOG("COM was already initialized in MTA");
}
if (SUCCEEDED(result)) {
CoUninitialize();
}
}
bool ok() {
return result == RPC_E_CHANGED_MODE || SUCCEEDED(result);
}
private:
HRESULT result;
};
typedef HANDLE (WINAPI *set_mm_thread_characteristics_function)(
const char * TaskName, LPDWORD TaskIndex);
typedef BOOL (WINAPI *revert_mm_thread_characteristics_function)(HANDLE handle);
extern cubeb_ops const wasapi_ops;
int wasapi_stream_stop(cubeb_stream * stm);
int wasapi_stream_start(cubeb_stream * stm);
void close_wasapi_stream(cubeb_stream * stm);
int setup_wasapi_stream(cubeb_stream * stm);
static char * wstr_to_utf8(const wchar_t * str);
static std::unique_ptr<const wchar_t[]> utf8_to_wstr(char* str);
}
struct cubeb
{
cubeb_ops const * ops;
/* Library dynamically opened to increase the render thread priority, and
the two function pointers we need. */
HMODULE mmcss_module;
set_mm_thread_characteristics_function set_mm_thread_characteristics;
revert_mm_thread_characteristics_function revert_mm_thread_characteristics;
};
class wasapi_endpoint_notification_client;
/* We have three possible callbacks we can use with a stream:
* - input only
* - output only
* - synchronized input and output
*
* Returns true when we should continue to play, false otherwise.
*/
typedef bool (*wasapi_refill_callback)(cubeb_stream * stm);
struct cubeb_stream
{
cubeb * context;
/* 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_stream_params output_mix_params;
/* Stream parameters. This is what the client requested,
* and what will be presented in the callback. */
cubeb_stream_params input_stream_params;
cubeb_stream_params output_stream_params;
/* The input and output device, or NULL for default. */
cubeb_devid input_device;
cubeb_devid output_device;
/* The latency initially requested for this stream, in frames. */
unsigned latency;
cubeb_state_callback state_callback;
cubeb_data_callback data_callback;
wasapi_refill_callback refill_callback;
void * user_ptr;
/* Lifetime considerations:
- client, render_client, audio_clock and audio_stream_volume are interface
pointer to the IAudioClient.
- The lifetime for device_enumerator and notification_client, resampler,
mix_buffer are the same as the cubeb_stream instance. */
/* Main handle on the WASAPI stream. */
IAudioClient * output_client;
/* Interface pointer to use the event-driven interface. */
IAudioRenderClient * render_client;
/* Interface pointer to use the volume facilities. */
IAudioStreamVolume * audio_stream_volume;
/* Interface pointer to use the stream audio clock. */
IAudioClock * audio_clock;
/* Frames written to the stream since it was opened. Reset on device
change. Uses mix_params.rate. */
UINT64 frames_written;
/* Frames written to the (logical) stream since it was first
created. Updated on device change. Uses stream_params.rate. */
UINT64 total_frames_written;
/* Last valid reported stream position. Used to ensure the position
reported by stream_get_position increases monotonically. */
UINT64 prev_position;
/* Device enumerator to be able to be notified when the default
device change. */
IMMDeviceEnumerator * device_enumerator;
/* Device notification client, to be able to be notified when the default
audio device changes and route the audio to the new default audio output
device */
wasapi_endpoint_notification_client * notification_client;
/* Main andle to the WASAPI capture stream. */
IAudioClient * input_client;
/* Interface to use the event driven capture interface */
IAudioCaptureClient * capture_client;
/* This event is set by the stream_stop and stream_destroy
function, so the render loop can exit properly. */
HANDLE shutdown_event;
/* Set by OnDefaultDeviceChanged when a stream reconfiguration is required.
The reconfiguration is handled by the render loop thread. */
HANDLE reconfigure_event;
/* This is set by WASAPI when we should refill the stream. */
HANDLE refill_event;
/* This is set by WASAPI when we should read from the input stream. In
* practice, we read from the input stream in the output callback, so
* this is not used, but it is necessary to start getting input data. */
HANDLE input_available_event;
/* Each cubeb_stream has its own thread. */
HANDLE thread;
/* The lock protects all members that are touched by the render thread or
change during a device reset, including: audio_clock, audio_stream_volume,
client, frames_written, mix_params, total_frames_written, prev_position. */
owned_critical_section stream_reset_lock;
/* Maximum number of frames that can be passed down in a callback. */
uint32_t input_buffer_frame_count;
/* Maximum number of frames that can be requested in a callback. */
uint32_t output_buffer_frame_count;
/* Resampler instance. Resampling will only happen if necessary. */
cubeb_resampler * resampler;
/* A buffer for up/down mixing multi-channel audio. */
float * mix_buffer;
/* WASAPI input works in "packets". We re-linearize the audio packets
* into this buffer before handing it to the resampler. */
auto_array<float> linear_input_buffer;
/* Stream volume. Set via stream_set_volume and used to reset volume on
device changes. */
float volume;
/* True if the stream is draining. */
bool draining;
/* True when we've destroyed the stream. This pointer is leaked on stream
* destruction if we could not join the thread. */
std::atomic<std::atomic<bool>*> emergency_bailout;
};
class wasapi_endpoint_notification_client : public IMMNotificationClient
{
public:
/* The implementation of MSCOM was copied from MSDN. */
ULONG STDMETHODCALLTYPE
AddRef()
{
return InterlockedIncrement(&ref_count);
}
ULONG STDMETHODCALLTYPE
Release()
{
ULONG ulRef = InterlockedDecrement(&ref_count);
if (0 == ulRef) {
delete this;
}
return ulRef;
}
HRESULT STDMETHODCALLTYPE
QueryInterface(REFIID riid, VOID **ppvInterface)
{
if (__uuidof(IUnknown) == riid) {
AddRef();
*ppvInterface = (IUnknown*)this;
} else if (__uuidof(IMMNotificationClient) == riid) {
AddRef();
*ppvInterface = (IMMNotificationClient*)this;
} else {
*ppvInterface = NULL;
return E_NOINTERFACE;
}
return S_OK;
}
wasapi_endpoint_notification_client(HANDLE event)
: ref_count(1)
, reconfigure_event(event)
{ }
virtual ~wasapi_endpoint_notification_client()
{ }
HRESULT STDMETHODCALLTYPE
OnDefaultDeviceChanged(EDataFlow flow, ERole role, LPCWSTR device_id)
{
LOG("Audio device default changed.");
/* we only support a single stream type for now. */
if (flow != eRender && role != eConsole) {
return S_OK;
}
BOOL ok = SetEvent(reconfigure_event);
if (!ok) {
LOG("SetEvent on reconfigure_event failed: %x", GetLastError());
}
return S_OK;
}
/* The remaining methods are not implemented, they simply log when called (if
log is enabled), for debugging. */
HRESULT STDMETHODCALLTYPE OnDeviceAdded(LPCWSTR device_id)
{
LOG("Audio device added.");
return S_OK;
};
HRESULT STDMETHODCALLTYPE OnDeviceRemoved(LPCWSTR device_id)
{
LOG("Audio device removed.");
return S_OK;
}
HRESULT STDMETHODCALLTYPE
OnDeviceStateChanged(LPCWSTR device_id, DWORD new_state)
{
LOG("Audio device state changed.");
return S_OK;
}
HRESULT STDMETHODCALLTYPE
OnPropertyValueChanged(LPCWSTR device_id, const PROPERTYKEY key)
{
LOG("Audio device property value changed.");
return S_OK;
}
private:
/* refcount for this instance, necessary to implement MSCOM semantics. */
LONG ref_count;
HANDLE reconfigure_event;
};
namespace {
bool has_input(cubeb_stream * stm)
{
return stm->input_stream_params.rate != 0;
}
bool has_output(cubeb_stream * stm)
{
return stm->output_stream_params.rate != 0;
}
bool should_upmix(cubeb_stream_params & stream, cubeb_stream_params & mixer)
{
return mixer.channels > stream.channels;
}
bool should_downmix(cubeb_stream_params & stream, cubeb_stream_params & mixer)
{
return mixer.channels < stream.channels;
}
double stream_to_mix_samplerate_ratio(cubeb_stream_params & stream, cubeb_stream_params & mixer)
{
return double(stream.rate) / mixer.rate;
}
uint32_t
get_rate(cubeb_stream * stm)
{
return has_input(stm) ? stm->input_stream_params.rate
: stm->output_stream_params.rate;
}
uint32_t
ms_to_hns(uint32_t ms)
{
return ms * 10000;
}
uint32_t
hns_to_ms(REFERENCE_TIME hns)
{
return static_cast<uint32_t>(hns / 10000);
}
double
hns_to_s(REFERENCE_TIME hns)
{
return static_cast<double>(hns) / 10000000;
}
uint32_t
hns_to_frames(cubeb_stream * stm, REFERENCE_TIME hns)
{
return hns_to_ms(hns * get_rate(stm)) / 1000;
}
uint32_t
hns_to_frames(uint32_t rate, REFERENCE_TIME hns)
{
return hns_to_ms(hns * rate) / 1000;
}
REFERENCE_TIME
frames_to_hns(cubeb_stream * stm, uint32_t frames)
{
return frames * 1000 / get_rate(stm);
}
/* Upmix function, copies a mono channel into L and R */
template<typename T>
void
mono_to_stereo(T * in, long insamples, T * out, int32_t out_channels)
{
for (int i = 0, j = 0; i < insamples; ++i, j += out_channels) {
out[j] = out[j + 1] = in[i];
}
}
template<typename T>
void
upmix(T * in, long inframes, T * out, int32_t in_channels, int32_t out_channels)
{
XASSERT(out_channels >= in_channels && in_channels > 0);
/* Either way, if we have 2 or more channels, the first two are L and R. */
/* If we are playing a mono stream over stereo speakers, copy the data over. */
if (in_channels == 1 && out_channels >= 2) {
mono_to_stereo(in, inframes, out, out_channels);
} else {
/* Copy through. */
for (int i = 0, o = 0; i < inframes * in_channels;
i += in_channels, o += out_channels) {
for (int j = 0; j < in_channels; ++j) {
out[o + j] = in[i + j];
}
}
}
/* Check if more channels. */
if (out_channels <= 2) {
return;
}
/* Put silence in remaining channels. */
for (long i = 0, o = 0; i < inframes; ++i, o += out_channels) {
for (int j = 2; j < out_channels; ++j) {
out[o + j] = 0.0;
}
}
}
template<typename T>
void
downmix(T * in, long inframes, T * out, int32_t in_channels, int32_t out_channels)
{
XASSERT(in_channels >= out_channels);
/* We could use a downmix matrix here, applying mixing weight based on the
channel, but directsound and winmm simply drop the channels that cannot be
rendered by the hardware, so we do the same for consistency. */
long out_index = 0;
for (long i = 0; i < inframes * in_channels; i += in_channels) {
for (int j = 0; j < out_channels; ++j) {
out[out_index + j] = in[i + j];
}
out_index += out_channels;
}
}
/* This returns the size of a frame in the stream, before the eventual upmix
occurs. */
static size_t
frames_to_bytes_before_mix(cubeb_stream * stm, size_t frames)
{
size_t stream_frame_size = stm->output_stream_params.channels * sizeof(float);
return stream_frame_size * frames;
}
/* This function handles the processing of the input and output audio,
* converting it to rate and channel layout specified at initialization.
* It then calls the data callback, via the resampler. */
long
refill(cubeb_stream * stm, float * input_buffer, long input_frames_count,
float * output_buffer, long output_frames_needed)
{
/* If we need to upmix after resampling, resample into the mix buffer to
avoid a copy. */
float * dest = nullptr;
if (has_output(stm)) {
if (should_upmix(stm->output_stream_params, stm->output_mix_params) ||
should_downmix(stm->output_stream_params, stm->output_mix_params)) {
dest = stm->mix_buffer;
} else {
dest = output_buffer;
}
}
long out_frames = cubeb_resampler_fill(stm->resampler,
input_buffer,
&input_frames_count,
dest,
output_frames_needed);
/* TODO: Report out_frames < 0 as an error via the API. */
XASSERT(out_frames >= 0);
{
auto_lock lock(stm->stream_reset_lock);
stm->frames_written += out_frames;
}
/* Go in draining mode if we got fewer frames than requested. */
if (out_frames < output_frames_needed) {
LOG("start draining.");
stm->draining = true;
}
/* If this is not true, there will be glitches.
It is alright to have produced less frames if we are draining, though. */
XASSERT(out_frames == output_frames_needed || stm->draining || !has_output(stm));
if (has_output(stm)) {
if (should_upmix(stm->output_stream_params, stm->output_mix_params)) {
upmix(dest, out_frames, output_buffer,
stm->output_stream_params.channels, stm->output_mix_params.channels);
} else if (should_downmix(stm->output_stream_params, stm->output_mix_params)) {
downmix(dest, out_frames, output_buffer,
stm->output_stream_params.channels, stm->output_mix_params.channels);
}
}
return out_frames;
}
/* This helper grabs all the frames available from a capture client, put them in
* linear_input_buffer. linear_input_buffer should be cleared before the
* callback exits. */
bool get_input_buffer(cubeb_stream * stm)
{
HRESULT hr;
UINT32 padding_in;
XASSERT(has_input(stm));
hr = stm->input_client->GetCurrentPadding(&padding_in);
if (FAILED(hr)) {
LOG("Failed to get padding");
return false;
}
XASSERT(padding_in <= stm->input_buffer_frame_count);
UINT32 total_available_input = padding_in;
BYTE * input_packet = NULL;
DWORD flags;
UINT64 dev_pos;
UINT32 next;
/* Get input packets until we have captured enough frames, and put them in a
* contiguous buffer. */
uint32_t offset = 0;
while (offset != total_available_input) {
hr = stm->capture_client->GetNextPacketSize(&next);
if (FAILED(hr)) {
LOG("cannot get next packet size: %x", hr);
return false;
}
/* This can happen if the capture stream has stopped. Just return in this
* case. */
if (!next) {
break;
}
UINT32 packet_size;
hr = stm->capture_client->GetBuffer(&input_packet,
&packet_size,
&flags,
&dev_pos,
NULL);
if (FAILED(hr)) {
LOG("GetBuffer failed for capture: %x", hr);
return false;
}
XASSERT(packet_size == next);
if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
LOG("insert silence: ps=%u", packet_size);
stm->linear_input_buffer.push_silence(packet_size * stm->input_stream_params.channels);
} else {
if (should_upmix(stm->input_mix_params, stm->input_stream_params)) {
bool ok = stm->linear_input_buffer.reserve(stm->linear_input_buffer.length() +
packet_size * stm->input_stream_params.channels);
assert(ok);
upmix(reinterpret_cast<float*>(input_packet), packet_size,
stm->linear_input_buffer.data() + stm->linear_input_buffer.length(),
stm->input_mix_params.channels,
stm->input_stream_params.channels);
stm->linear_input_buffer.set_length(stm->linear_input_buffer.length() + packet_size * stm->input_stream_params.channels);
} else if (should_downmix(stm->input_mix_params, stm->input_stream_params)) {
bool ok = stm->linear_input_buffer.reserve(stm->linear_input_buffer.length() +
packet_size * stm->input_stream_params.channels);
assert(ok);
downmix(reinterpret_cast<float*>(input_packet), packet_size,
stm->linear_input_buffer.data() + stm->linear_input_buffer.length(),
stm->input_mix_params.channels,
stm->input_stream_params.channels);
stm->linear_input_buffer.set_length(stm->linear_input_buffer.length() + packet_size * stm->input_stream_params.channels);
} else {
stm->linear_input_buffer.push(reinterpret_cast<float*>(input_packet),
packet_size * stm->input_stream_params.channels);
}
}
hr = stm->capture_client->ReleaseBuffer(packet_size);
if (FAILED(hr)) {
LOG("FAILED to release intput buffer");
return false;
}
offset += packet_size;
}
assert(stm->linear_input_buffer.length() >= total_available_input &&
offset == total_available_input);
return true;
}
/* Get an output buffer from the render_client. It has to be released before
* exiting the callback. */
bool get_output_buffer(cubeb_stream * stm, float *& buffer, size_t & frame_count)
{
UINT32 padding_out;
HRESULT hr;
XASSERT(has_output(stm));
hr = stm->output_client->GetCurrentPadding(&padding_out);
if (FAILED(hr)) {
LOG("Failed to get padding: %x", hr);
return false;
}
XASSERT(padding_out <= stm->output_buffer_frame_count);
if (stm->draining) {
if (padding_out == 0) {
LOG("Draining finished.");
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED);
return false;
}
LOG("Draining.");
return true;
}
frame_count = stm->output_buffer_frame_count - padding_out;
BYTE * output_buffer;
hr = stm->render_client->GetBuffer(frame_count, &output_buffer);
if (FAILED(hr)) {
LOG("cannot get render buffer");
return false;
}
buffer = reinterpret_cast<float*>(output_buffer);
return true;
}
/**
* This function gets input data from a input device, and pass it along with an
* output buffer to the resamplers. */
bool
refill_callback_duplex(cubeb_stream * stm)
{
HRESULT hr;
float * output_buffer = nullptr;
size_t output_frames = 0;
size_t input_frames;
bool rv;
XASSERT(has_input(stm) && has_output(stm));
rv = get_input_buffer(stm);
if (!rv) {
return rv;
}
input_frames = stm->linear_input_buffer.length() / stm->input_stream_params.channels;
if (!input_frames) {
return true;
}
rv = get_output_buffer(stm, output_buffer, output_frames);
if (!rv) {
hr = stm->render_client->ReleaseBuffer(output_frames, 0);
return rv;
}
/* This can only happen when debugging, and having breakpoints set in the
* callback in a way that it makes the stream underrun. */
if (output_frames == 0) {
return true;
}
// When WASAPI has not filled the input buffer yet, send silence.
double output_duration = double(output_frames) / stm->output_mix_params.rate;
double input_duration = double(stm->linear_input_buffer.length() / stm->input_stream_params.channels) / stm->input_mix_params.rate;
if (input_duration < output_duration) {
size_t padding = size_t(round((output_duration - input_duration) * stm->input_mix_params.rate));
LOG("padding silence: out=%f in=%f pad=%u", output_duration, input_duration, padding);
stm->linear_input_buffer.push_front_silence(padding * stm->input_stream_params.channels);
}
LOGV("Duplex callback: input frames: %zu, output frames: %zu",
stm->linear_input_buffer.length(), output_frames);
refill(stm,
stm->linear_input_buffer.data(),
stm->linear_input_buffer.length(),
output_buffer,
output_frames);
stm->linear_input_buffer.clear();
hr = stm->render_client->ReleaseBuffer(output_frames, 0);
if (FAILED(hr)) {
LOG("failed to release buffer: %x", hr);
return false;
}
return true;
}
bool
refill_callback_input(cubeb_stream * stm)
{
bool rv, consumed_all_buffer;
XASSERT(has_input(stm) && !has_output(stm));
rv = get_input_buffer(stm);
if (!rv) {
return rv;
}
// This can happen at the very beginning of the stream.
if (!stm->linear_input_buffer.length()) {
return true;
}
LOGV("Input callback: input frames: %zu", stm->linear_input_buffer.length());
long read = refill(stm,
stm->linear_input_buffer.data(),
stm->linear_input_buffer.length(),
nullptr,
0);
consumed_all_buffer = read == stm->linear_input_buffer.length();
stm->linear_input_buffer.clear();
return consumed_all_buffer;
}
bool
refill_callback_output(cubeb_stream * stm)
{
bool rv;
HRESULT hr;
float * output_buffer = nullptr;
size_t output_frames = 0;
XASSERT(!has_input(stm) && has_output(stm));
rv = get_output_buffer(stm, output_buffer, output_frames);
if (!rv) {
return rv;
}
if (stm->draining || output_frames == 0) {
return true;
}
long got = refill(stm,
nullptr,
0,
output_buffer,
output_frames);
LOGV("Output callback: output frames requested: %zu, got %ld",
output_frames, got);
XASSERT(got >= 0);
XASSERT(got == output_frames || stm->draining);
hr = stm->render_client->ReleaseBuffer(got, 0);
if (FAILED(hr)) {
LOG("failed to release buffer: %x", hr);
return false;
}
return got == output_frames || stm->draining;
}
static unsigned int __stdcall
wasapi_stream_render_loop(LPVOID stream)
{
cubeb_stream * stm = static_cast<cubeb_stream *>(stream);
std::atomic<bool> * emergency_bailout = stm->emergency_bailout;
bool is_playing = true;
HANDLE wait_array[4] = {
stm->shutdown_event,
stm->reconfigure_event,
stm->refill_event,
stm->input_available_event
};
HANDLE mmcss_handle = NULL;
HRESULT hr = 0;
DWORD mmcss_task_index = 0;
auto_com com;
if (!com.ok()) {
LOG("COM initialization failed on render_loop thread.");
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR);
return 0;
}
/* We could consider using "Pro Audio" here for WebAudio and
maybe WebRTC. */
mmcss_handle =
stm->context->set_mm_thread_characteristics("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: %x", GetLastError());
}
// This has already been nulled out, simply exit.
if (!emergency_bailout) {
is_playing = false;
}
/* WaitForMultipleObjects timeout can trigger in cases where we don't want to
treat it as a timeout, such as across a system sleep/wake cycle. Trigger
the timeout error handling only when the timeout_limit is reached, which is
reset on each successful loop. */
unsigned timeout_count = 0;
const unsigned timeout_limit = 5;
while (is_playing) {
// We want to check the emergency bailout variable before a
// and after the WaitForMultipleObject, because the handles WaitForMultipleObjects
// is going to wait on might have been closed already.
if (*emergency_bailout) {
delete emergency_bailout;
return 0;
}
DWORD waitResult = WaitForMultipleObjects(ARRAY_LENGTH(wait_array),
wait_array,
FALSE,
1000);
if (*emergency_bailout) {
delete emergency_bailout;
return 0;
}
if (waitResult != WAIT_TIMEOUT) {
timeout_count = 0;
}
switch (waitResult) {
case WAIT_OBJECT_0: { /* shutdown */
is_playing = false;
/* We don't check if the drain is actually finished here, we just want to
shutdown. */
if (stm->draining) {
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_DRAINED);
}
continue;
}
case WAIT_OBJECT_0 + 1: { /* reconfigure */
XASSERT(stm->output_client || stm->input_client);
LOG("Reconfiguring the stream");
/* Close the stream */
if (stm->output_client) {
stm->output_client->Stop();
LOG("Output stopped.");
}
if (stm->input_client) {
stm->input_client->Stop();
LOG("Input stopped.");
}
{
auto_lock lock(stm->stream_reset_lock);
close_wasapi_stream(stm);
LOG("Stream closed.");
/* Reopen a stream and start it immediately. This will automatically pick the
new default device for this role. */
int r = setup_wasapi_stream(stm);
if (r != CUBEB_OK) {
LOG("Error setting up the stream during reconfigure.");
/* Don't destroy the stream here, since we expect the caller to do
so after the error has propagated via the state callback. */
is_playing = false;
hr = E_FAIL;
continue;
}
LOG("Stream setup successfuly.");
}
XASSERT(stm->output_client || stm->input_client);
if (stm->output_client) {
stm->output_client->Start();
LOG("Output started after reconfigure.");
}
if (stm->input_client) {
stm->input_client->Start();
LOG("Input started after reconfigure.");
}
break;
}
case WAIT_OBJECT_0 + 2: /* refill */
XASSERT(has_input(stm) && has_output(stm) ||
!has_input(stm) && has_output(stm));
is_playing = stm->refill_callback(stm);
break;
case WAIT_OBJECT_0 + 3: /* input available */
if (has_input(stm) && has_output(stm)) { continue; }
is_playing = stm->refill_callback(stm);
break;
case WAIT_TIMEOUT:
XASSERT(stm->shutdown_event == wait_array[0]);
if (++timeout_count >= timeout_limit) {
LOG("Render loop reached the timeout limit.");
is_playing = false;
hr = E_FAIL;
}
break;
default:
LOG("case %d not handled in render loop.", waitResult);
abort();
}
}
if (FAILED(hr)) {
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_ERROR);
}
stm->context->revert_mm_thread_characteristics(mmcss_handle);
return 0;
}
void wasapi_destroy(cubeb * context);
HANDLE WINAPI set_mm_thread_characteristics_noop(const char *, 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)
{
HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator),
NULL, CLSCTX_INPROC_SERVER,
IID_PPV_ARGS(&stm->device_enumerator));
if (FAILED(hr)) {
LOG("Could not get device enumerator: %x", hr);
return hr;
}
stm->notification_client = new wasapi_endpoint_notification_client(stm->reconfigure_event);
hr = stm->device_enumerator->RegisterEndpointNotificationCallback(stm->notification_client);
if (FAILED(hr)) {
LOG("Could not register endpoint notification callback: %x", hr);
SafeRelease(stm->notification_client);
stm->notification_client = nullptr;
SafeRelease(stm->device_enumerator);
stm->device_enumerator = nullptr;
}
return hr;
}
HRESULT unregister_notification_client(cubeb_stream * stm)
{
XASSERT(stm);
HRESULT hr;
if (!stm->device_enumerator) {
return S_OK;
}
hr = stm->device_enumerator->UnregisterEndpointNotificationCallback(stm->notification_client);
if (FAILED(hr)) {
// We can't really do anything here, we'll probably leak the
// notification client, but we can at least release the enumerator.
SafeRelease(stm->device_enumerator);
return S_OK;
}
SafeRelease(stm->notification_client);
SafeRelease(stm->device_enumerator);
return S_OK;
}
HRESULT get_endpoint(IMMDevice ** device, LPCWSTR devid)
{
IMMDeviceEnumerator * enumerator;
HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator),
NULL, CLSCTX_INPROC_SERVER,
IID_PPV_ARGS(&enumerator));
if (FAILED(hr)) {
LOG("Could not get device enumerator: %x", hr);
return hr;
}
hr = enumerator->GetDevice(devid, device);
if (FAILED(hr)) {
LOG("Could not get device: %x", hr);
SafeRelease(enumerator);
return hr;
}
SafeRelease(enumerator);
return S_OK;
}
HRESULT get_default_endpoint(IMMDevice ** device, EDataFlow direction)
{
IMMDeviceEnumerator * enumerator;
HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator),
NULL, CLSCTX_INPROC_SERVER,
IID_PPV_ARGS(&enumerator));
if (FAILED(hr)) {
LOG("Could not get device enumerator: %x", hr);
return hr;
}
hr = enumerator->GetDefaultAudioEndpoint(direction, eConsole, device);
if (FAILED(hr)) {
LOG("Could not get default audio endpoint: %x", hr);
SafeRelease(enumerator);
return hr;
}
SafeRelease(enumerator);
return ERROR_SUCCESS;
}
double
current_stream_delay(cubeb_stream * stm)
{
stm->stream_reset_lock.assert_current_thread_owns();
/* If the default audio endpoint went away during playback and we weren't
able to configure a new one, it's possible the caller may call this
before the error callback has propogated back. */
if (!stm->audio_clock) {
return 0;
}
UINT64 freq;
HRESULT hr = stm->audio_clock->GetFrequency(&freq);
if (FAILED(hr)) {
LOG("GetFrequency failed: %x", hr);
return 0;
}
UINT64 pos;
hr = stm->audio_clock->GetPosition(&pos, NULL);
if (FAILED(hr)) {
LOG("GetPosition failed: %x", hr);
return 0;
}
double cur_pos = static_cast<double>(pos) / freq;
double max_pos = static_cast<double>(stm->frames_written) / stm->output_mix_params.rate;
double delay = max_pos - cur_pos;
XASSERT(delay >= 0);
return delay;
}
int
stream_set_volume(cubeb_stream * stm, float volume)
{
stm->stream_reset_lock.assert_current_thread_owns();
if (!stm->audio_stream_volume) {
return CUBEB_ERROR;
}
uint32_t channels;
HRESULT hr = stm->audio_stream_volume->GetChannelCount(&channels);
if (hr != S_OK) {
LOG("could not get the channel count: %x", hr);
return CUBEB_ERROR;
}
/* up to 9.1 for now */
if (channels > 10) {
return CUBEB_ERROR_NOT_SUPPORTED;
}
float volumes[10];
for (uint32_t i = 0; i < channels; i++) {
volumes[i] = volume;
}
hr = stm->audio_stream_volume->SetAllVolumes(channels, volumes);
if (hr != S_OK) {
LOG("could not set the channels volume: %x", hr);
return CUBEB_ERROR;
}
return CUBEB_OK;
}
} // namespace anonymous
extern "C" {
int wasapi_init(cubeb ** context, char const * context_name)
{
HRESULT hr;
auto_com com;
if (!com.ok()) {
return CUBEB_ERROR;
}
/* We don't use the device yet, but need to make sure we can initialize one
so that this backend is not incorrectly enabled on platforms that don't
support WASAPI. */
IMMDevice * device;
hr = get_default_endpoint(&device, eRender);
if (FAILED(hr)) {
LOG("Could not get device: %x", hr);
return CUBEB_ERROR;
}
SafeRelease(device);
cubeb * ctx = (cubeb *)calloc(1, sizeof(cubeb));
if (!ctx) {
return CUBEB_ERROR;
}
ctx->ops = &wasapi_ops;
ctx->mmcss_module = LoadLibraryA("Avrt.dll");
if (ctx->mmcss_module) {
ctx->set_mm_thread_characteristics =
(set_mm_thread_characteristics_function) GetProcAddress(
ctx->mmcss_module, "AvSetMmThreadCharacteristicsA");
ctx->revert_mm_thread_characteristics =
(revert_mm_thread_characteristics_function) GetProcAddress(
ctx->mmcss_module, "AvRevertMmThreadCharacteristics");
if (!(ctx->set_mm_thread_characteristics && ctx->revert_mm_thread_characteristics)) {
LOG("Could not load AvSetMmThreadCharacteristics or AvRevertMmThreadCharacteristics: %x", GetLastError());
FreeLibrary(ctx->mmcss_module);
}
} else {
// 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");
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;
}
}
namespace {
bool stop_and_join_render_thread(cubeb_stream * stm)
{
bool rv = true;
LOG("Stop and join render thread.");
if (!stm->thread) {
LOG("No thread present.");
return true;
}
// If we've already leaked the thread, just return,
// there is not much we can do.
if (!stm->emergency_bailout.load()) {
return false;
}
BOOL ok = SetEvent(stm->shutdown_event);
if (!ok) {
LOG("Destroy SetEvent failed: %d", GetLastError());
}
/* Wait five seconds for the rendering thread to return. It's supposed to
* check its event loop very often, five seconds is rather conservative. */
DWORD r = WaitForSingleObject(stm->thread, 5000);
if (r == WAIT_TIMEOUT) {
/* Something weird happened, leak the thread and continue the shutdown
* process. */
*(stm->emergency_bailout) = true;
// We give the ownership to the rendering thread.
stm->emergency_bailout = nullptr;
LOG("Destroy WaitForSingleObject on thread timed out,"
" leaking the thread: %d", GetLastError());
rv = false;
}
if (r == WAIT_FAILED) {
*(stm->emergency_bailout) = true;
// We give the ownership to the rendering thread.
stm->emergency_bailout = nullptr;
LOG("Destroy WaitForSingleObject on thread failed: %d", GetLastError());
rv = false;
}
// Only attempts to close and null out the thread and event if the
// WaitForSingleObject above succeeded, so that calling this function again
// attemps to clean up the thread and event each time.
if (rv) {
LOG("Closing thread.");
CloseHandle(stm->thread);
stm->thread = NULL;
CloseHandle(stm->shutdown_event);
stm->shutdown_event = 0;
}
return rv;
}
void wasapi_destroy(cubeb * context)
{
if (context->mmcss_module) {
FreeLibrary(context->mmcss_module);
}
free(context);
}
char const * wasapi_get_backend_id(cubeb * context)
{
return "wasapi";
}
int
wasapi_get_max_channel_count(cubeb * ctx, uint32_t * max_channels)
{
HRESULT hr;
IAudioClient * client;
WAVEFORMATEX * mix_format;
auto_com com;
if (!com.ok()) {
return CUBEB_ERROR;
}
XASSERT(ctx && max_channels);
IMMDevice * device;
hr = get_default_endpoint(&device, eRender);
if (FAILED(hr)) {
return CUBEB_ERROR;
}
hr = device->Activate(__uuidof(IAudioClient),
CLSCTX_INPROC_SERVER,
NULL, (void **)&client);
SafeRelease(device);
if (FAILED(hr)) {
return CUBEB_ERROR;
}
hr = client->GetMixFormat(&mix_format);
if (FAILED(hr)) {
SafeRelease(client);
return CUBEB_ERROR;
}
*max_channels = mix_format->nChannels;
CoTaskMemFree(mix_format);
SafeRelease(client);
return CUBEB_OK;
}
int
wasapi_get_min_latency(cubeb * ctx, cubeb_stream_params params, uint32_t * latency_frames)
{
HRESULT hr;
IAudioClient * client;
REFERENCE_TIME default_period;
auto_com com;
if (!com.ok()) {
return CUBEB_ERROR;
}
if (params.format != CUBEB_SAMPLE_FLOAT32NE) {
return CUBEB_ERROR_INVALID_FORMAT;
}
IMMDevice * device;
hr = get_default_endpoint(&device, eRender);
if (FAILED(hr)) {
LOG("Could not get default endpoint: %x", hr);
return CUBEB_ERROR;
}
hr = device->Activate(__uuidof(IAudioClient),
CLSCTX_INPROC_SERVER,
NULL, (void **)&client);
SafeRelease(device);
if (FAILED(hr)) {
LOG("Could not activate device for latency: %x", hr);
return CUBEB_ERROR;
}
/* The second parameter is for exclusive mode, that we don't use. */
hr = client->GetDevicePeriod(&default_period, NULL);
if (FAILED(hr)) {
SafeRelease(client);
LOG("Could not get device period: %x", hr);
return CUBEB_ERROR;
}
LOG("default device period: %lld", default_period);
/* 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 */
*latency_frames = hns_to_frames(params.rate, default_period);
LOG("Minimum latency in frames: %u", *latency_frames);
SafeRelease(client);
return CUBEB_OK;
}
int
wasapi_get_preferred_sample_rate(cubeb * ctx, uint32_t * rate)
{
HRESULT hr;
IAudioClient * client;
WAVEFORMATEX * mix_format;
auto_com com;
if (!com.ok()) {
return CUBEB_ERROR;
}
IMMDevice * device;
hr = get_default_endpoint(&device, eRender);
if (FAILED(hr)) {
return CUBEB_ERROR;
}
hr = device->Activate(__uuidof(IAudioClient),
CLSCTX_INPROC_SERVER,
NULL, (void **)&client);
SafeRelease(device);
if (FAILED(hr)) {
return CUBEB_ERROR;
}
hr = client->GetMixFormat(&mix_format);
if (FAILED(hr)) {
SafeRelease(client);
return CUBEB_ERROR;
}
*rate = mix_format->nSamplesPerSec;
LOG("Preferred sample rate for output: %u", *rate);
CoTaskMemFree(mix_format);
SafeRelease(client);
return CUBEB_OK;
}
void wasapi_stream_destroy(cubeb_stream * stm);
/* Based on the mix format and the stream format, try to find a way to play
what the user requested. */
static void
handle_channel_layout(cubeb_stream * stm, WAVEFORMATEX ** mix_format, const cubeb_stream_params * stream_params)
{
/* Common case: the hardware is stereo. Up-mixing and down-mixing will be
handled in the callback. */
if ((*mix_format)->nChannels <= 2) {
return;
}
/* The docs say that GetMixFormat is always of type WAVEFORMATEXTENSIBLE [1],
so the reinterpret_cast below should be safe. In practice, this is not
true, and we just want to bail out and let the rest of the code find a good
conversion path instead of trying to make WASAPI do it by itself.
[1]: http://msdn.microsoft.com/en-us/library/windows/desktop/dd370811%28v=vs.85%29.aspx*/
if ((*mix_format)->wFormatTag != WAVE_FORMAT_EXTENSIBLE) {
return;
}
WAVEFORMATEXTENSIBLE * format_pcm = reinterpret_cast<WAVEFORMATEXTENSIBLE *>(*mix_format);
/* Stash a copy of the original mix format in case we need to restore it later. */
WAVEFORMATEXTENSIBLE hw_mix_format = *format_pcm;
/* The hardware is in surround mode, we want to only use front left and front
right. Try that, and check if it works. */
switch (stream_params->channels) {
case 1: /* Mono */
format_pcm->dwChannelMask = KSAUDIO_SPEAKER_MONO;
break;
case 2: /* Stereo */
format_pcm->dwChannelMask = KSAUDIO_SPEAKER_STEREO;
break;
default:
XASSERT(false && "Channel layout not supported.");
break;
}
(*mix_format)->nChannels = stream_params->channels;
(*mix_format)->nBlockAlign = ((*mix_format)->wBitsPerSample * (*mix_format)->nChannels) / 8;
(*mix_format)->nAvgBytesPerSec = (*mix_format)->nSamplesPerSec * (*mix_format)->nBlockAlign;
format_pcm->SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
(*mix_format)->wBitsPerSample = 32;
format_pcm->Samples.wValidBitsPerSample = (*mix_format)->wBitsPerSample;
/* Check if wasapi will accept our channel layout request. */
WAVEFORMATEX * closest;
HRESULT hr = stm->output_client->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED,
*mix_format,
&closest);
if (hr == S_FALSE) {
/* Not supported, but WASAPI gives us a suggestion. Use it, and handle the
eventual upmix/downmix ourselves */
LOG("Using WASAPI suggested format: channels: %d", closest->nChannels);
WAVEFORMATEXTENSIBLE * closest_pcm = reinterpret_cast<WAVEFORMATEXTENSIBLE *>(closest);
XASSERT(closest_pcm->SubFormat == format_pcm->SubFormat);
CoTaskMemFree(*mix_format);
*mix_format = closest;
} else if (hr == AUDCLNT_E_UNSUPPORTED_FORMAT) {
/* Not supported, no suggestion. This should not happen, but it does in the
field with some sound cards. We restore the mix format, and let the rest
of the code figure out the right conversion path. */
*reinterpret_cast<WAVEFORMATEXTENSIBLE *>(*mix_format) = hw_mix_format;
} else if (hr == S_OK) {
LOG("Requested format accepted by WASAPI.");
} else {
LOG("IsFormatSupported unhandled error: %x", hr);
}
}
#define DIRECTION_NAME (direction == eCapture ? "capture" : "render")
template<typename T>
int setup_wasapi_stream_one_side(cubeb_stream * stm,
cubeb_stream_params * stream_params,
cubeb_devid devid,
EDataFlow direction,
REFIID riid,
IAudioClient ** audio_client,
uint32_t * buffer_frame_count,
HANDLE & event,
T ** render_or_capture_client,
cubeb_stream_params * mix_params)
{
IMMDevice * device;
WAVEFORMATEX * mix_format;
HRESULT hr;
stm->stream_reset_lock.assert_current_thread_owns();
bool try_again = false;
// This loops until we find a device that works, or we've exhausted all
// possibilities.
do {
if (devid) {
std::unique_ptr<const wchar_t[]> id(utf8_to_wstr(reinterpret_cast<char*>(devid)));
hr = get_endpoint(&device, id.get());
if (FAILED(hr)) {
LOG("Could not get %s endpoint, error: %x\n", DIRECTION_NAME, hr);
return CUBEB_ERROR;
}
}
else {
hr = get_default_endpoint(&device, direction);
if (FAILED(hr)) {
LOG("Could not get default %s endpoint, error: %x\n", DIRECTION_NAME, hr);
return CUBEB_ERROR;
}
}
/* Get a client. We will get all other interfaces we need from
* this pointer. */
hr = device->Activate(__uuidof(IAudioClient),
CLSCTX_INPROC_SERVER,
NULL, (void **)audio_client);
SafeRelease(device);
if (FAILED(hr)) {
LOG("Could not activate the device to get an audio"
" client for %s: error: %x\n", DIRECTION_NAME, hr);
// A particular device can't be activated because it has been
// unplugged, try fall back to the default audio device.
if (devid && hr == AUDCLNT_E_DEVICE_INVALIDATED) {
LOG("Trying again with the default %s audio device.", DIRECTION_NAME);
devid = nullptr;
try_again = true;
} else {
return CUBEB_ERROR;
}
} else {
try_again = false;
}
} while (try_again);
/* We have to distinguish between the format the mixer uses,
* and the format the stream we want to play uses. */
hr = (*audio_client)->GetMixFormat(&mix_format);
if (FAILED(hr)) {
LOG("Could not fetch current mix format from the audio"
" client for %s: error: %x", DIRECTION_NAME, hr);
return CUBEB_ERROR;
}
handle_channel_layout(stm, &mix_format, stream_params);
/* Shared mode WASAPI always supports float32 sample format, so this
* is safe. */
mix_params->format = CUBEB_SAMPLE_FLOAT32NE;
mix_params->rate = mix_format->nSamplesPerSec;
mix_params->channels = mix_format->nChannels;
LOG("Setup requested=[f=%d r=%u c=%u] mix=[f=%d r=%u c=%u]",
stream_params->format, stream_params->rate, stream_params->channels,
mix_params->format, mix_params->rate, mix_params->channels);
hr = (*audio_client)->Initialize(AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK |
AUDCLNT_STREAMFLAGS_NOPERSIST,
frames_to_hns(stm, stm->latency),
0,
mix_format,
NULL);
if (FAILED(hr)) {
LOG("Unable to initialize audio client for %s: %x.", DIRECTION_NAME, hr);
return CUBEB_ERROR;
}
CoTaskMemFree(mix_format);
hr = (*audio_client)->GetBufferSize(buffer_frame_count);
if (FAILED(hr)) {
LOG("Could not get the buffer size from the client"
" for %s %x.", DIRECTION_NAME, hr);
return CUBEB_ERROR;
}
// Input is up/down mixed when depacketized in get_input_buffer.
if (has_output(stm) &&
(should_upmix(*stream_params, *mix_params) ||
should_downmix(*stream_params, *mix_params))) {
stm->mix_buffer = (float *)malloc(frames_to_bytes_before_mix(stm, *buffer_frame_count));
}
hr = (*audio_client)->SetEventHandle(event);
if (FAILED(hr)) {
LOG("Could set the event handle for the %s client %x.",
DIRECTION_NAME, hr);
return CUBEB_ERROR;
}
hr = (*audio_client)->GetService(riid, (void **)render_or_capture_client);
if (FAILED(hr)) {
LOG("Could not get the %s client %x.", DIRECTION_NAME, hr);
return CUBEB_ERROR;
}
return CUBEB_OK;
}
#undef DIRECTION_NAME
int setup_wasapi_stream(cubeb_stream * stm)
{
HRESULT hr;
int rv;
stm->stream_reset_lock.assert_current_thread_owns();
auto_com com;
if (!com.ok()) {
LOG("Failure to initialize COM.");
return CUBEB_ERROR;
}
XASSERT((!stm->output_client || !stm->input_client) && "WASAPI stream already setup, close it first.");
if (has_input(stm)) {
LOG("Setup capture: device=%x", (int)stm->input_device);
rv = setup_wasapi_stream_one_side(stm,
&stm->input_stream_params,
stm->input_device,
eCapture,
__uuidof(IAudioCaptureClient),
&stm->input_client,
&stm->input_buffer_frame_count,
stm->input_available_event,
&stm->capture_client,
&stm->input_mix_params);
if (rv != CUBEB_OK) {
LOG("Failure to open the input side.");
return rv;
}
}
if (has_output(stm)) {
LOG("Setup render: device=%x", (int)stm->output_device);
rv = setup_wasapi_stream_one_side(stm,
&stm->output_stream_params,
stm->output_device,
eRender,
__uuidof(IAudioRenderClient),
&stm->output_client,
&stm->output_buffer_frame_count,
stm->refill_event,
&stm->render_client,
&stm->output_mix_params);
if (rv != CUBEB_OK) {
LOG("Failure to open the output side.");
return rv;
}
hr = stm->output_client->GetService(__uuidof(IAudioStreamVolume),
(void **)&stm->audio_stream_volume);
if (FAILED(hr)) {
LOG("Could not get the IAudioStreamVolume: %x", hr);
return CUBEB_ERROR;
}
XASSERT(stm->frames_written == 0);
hr = stm->output_client->GetService(__uuidof(IAudioClock),
(void **)&stm->audio_clock);
if (FAILED(hr)) {
LOG("Could not get the IAudioClock: %x", hr);
return CUBEB_ERROR;
}
/* Restore the stream volume over a device change. */
if (stream_set_volume(stm, stm->volume) != CUBEB_OK) {
LOG("Could not set the volume.");
return CUBEB_ERROR;
}
}
/* If we have both input and output, we resample to
* the highest sample rate available. */
int32_t target_sample_rate;
if (has_input(stm) && has_output(stm)) {
assert(stm->input_stream_params.rate == stm->output_stream_params.rate);
target_sample_rate = stm->input_stream_params.rate;
} else if (has_input(stm)) {
target_sample_rate = stm->input_stream_params.rate;
} else {
XASSERT(has_output(stm));
target_sample_rate = stm->output_stream_params.rate;
}
LOG("Target sample rate: %d", target_sample_rate);
/* If we are playing/capturing a mono stream, we only resample one channel,
and copy it over, so we are always resampling the number
of channels of the stream, not the number of channels
that WASAPI wants. */
cubeb_stream_params input_params = stm->input_mix_params;
input_params.channels = stm->input_stream_params.channels;
cubeb_stream_params output_params = stm->output_mix_params;
output_params.channels = stm->output_stream_params.channels;
stm->resampler =
cubeb_resampler_create(stm,
has_input(stm) ? &input_params : nullptr,
has_output(stm) ? &output_params : nullptr,
target_sample_rate,
stm->data_callback,
stm->user_ptr,
CUBEB_RESAMPLER_QUALITY_DESKTOP);
if (!stm->resampler) {
LOG("Could not get a resampler");
return CUBEB_ERROR;
}
XASSERT(has_input(stm) || has_output(stm));
if (has_input(stm) && has_output(stm)) {
stm->refill_callback = refill_callback_duplex;
} else if (has_input(stm)) {
stm->refill_callback = refill_callback_input;
} else if (has_output(stm)) {
stm->refill_callback = refill_callback_output;
}
return CUBEB_OK;
}
int
wasapi_stream_init(cubeb * context, cubeb_stream ** stream,
char const * stream_name,
cubeb_devid input_device,
cubeb_stream_params * input_stream_params,
cubeb_devid output_device,
cubeb_stream_params * output_stream_params,
unsigned int latency_frames, cubeb_data_callback data_callback,
cubeb_state_callback state_callback, void * user_ptr)
{
HRESULT hr;
int rv;
auto_com com;
if (!com.ok()) {
return CUBEB_ERROR;
}
XASSERT(context && stream && (input_stream_params || output_stream_params));
if (output_stream_params && output_stream_params->format != CUBEB_SAMPLE_FLOAT32NE ||
input_stream_params && input_stream_params->format != CUBEB_SAMPLE_FLOAT32NE) {
LOG("Invalid format, %p %p %d %d",
output_stream_params, input_stream_params,
output_stream_params && output_stream_params->format,
input_stream_params && input_stream_params->format);
return CUBEB_ERROR_INVALID_FORMAT;
}
cubeb_stream * stm = (cubeb_stream *)calloc(1, sizeof(cubeb_stream));
XASSERT(stm);
stm->context = context;
stm->data_callback = data_callback;
stm->state_callback = state_callback;
stm->user_ptr = user_ptr;
stm->draining = false;
if (input_stream_params) {
stm->input_stream_params = *input_stream_params;
stm->input_device = input_device;
}
if (output_stream_params) {
stm->output_stream_params = *output_stream_params;
stm->output_device = output_device;
}
stm->latency = latency_frames;
stm->volume = 1.0;
// Placement new to call ctor.
new (&stm->stream_reset_lock) owned_critical_section();
stm->reconfigure_event = CreateEvent(NULL, 0, 0, NULL);
if (!stm->reconfigure_event) {
LOG("Can't create the reconfigure event, error: %x", GetLastError());
wasapi_stream_destroy(stm);
return CUBEB_ERROR;
}
/* Unconditionally create the two events so that the wait logic is simpler. */
stm->refill_event = CreateEvent(NULL, 0, 0, NULL);
if (!stm->refill_event) {
LOG("Can't create the refill event, error: %x", GetLastError());
wasapi_stream_destroy(stm);
return CUBEB_ERROR;
}
stm->input_available_event = CreateEvent(NULL, 0, 0, NULL);
if (!stm->input_available_event) {
LOG("Can't create the input available event , error: %x", GetLastError());
wasapi_stream_destroy(stm);
return CUBEB_ERROR;
}
{
/* Locking here is not strictly necessary, because we don't have a
notification client that can reset the stream yet, but it lets us
assert that the lock is held in the function. */
auto_lock lock(stm->stream_reset_lock);
rv = setup_wasapi_stream(stm);
}
if (rv != CUBEB_OK) {
wasapi_stream_destroy(stm);
return rv;
}
hr = register_notification_client(stm);
if (FAILED(hr)) {
/* this is not fatal, we can still play audio, but we won't be able
to keep using the default audio endpoint if it changes. */
LOG("failed to register notification client, %x", hr);
}
*stream = stm;
return CUBEB_OK;
}
void close_wasapi_stream(cubeb_stream * stm)
{
XASSERT(stm);
stm->stream_reset_lock.assert_current_thread_owns();
SafeRelease(stm->output_client);
stm->output_client = NULL;
SafeRelease(stm->input_client);
stm->input_client = NULL;
SafeRelease(stm->render_client);
stm->render_client = NULL;
SafeRelease(stm->capture_client);
stm->capture_client = NULL;
SafeRelease(stm->audio_stream_volume);
stm->audio_stream_volume = NULL;
SafeRelease(stm->audio_clock);
stm->audio_clock = NULL;
stm->total_frames_written += static_cast<UINT64>(round(stm->frames_written * stream_to_mix_samplerate_ratio(stm->output_stream_params, stm->output_mix_params)));
stm->frames_written = 0;
if (stm->resampler) {
cubeb_resampler_destroy(stm->resampler);
stm->resampler = NULL;
}
free(stm->mix_buffer);
stm->mix_buffer = NULL;
}
void wasapi_stream_destroy(cubeb_stream * stm)
{
XASSERT(stm);
// Only free stm->emergency_bailout if we could not join the thread.
// If we could not join the thread, stm->emergency_bailout is true
// and is still alive until the thread wakes up and exits cleanly.
if (stop_and_join_render_thread(stm)) {
delete stm->emergency_bailout.load();
stm->emergency_bailout = nullptr;
}
unregister_notification_client(stm);
SafeRelease(stm->reconfigure_event);
SafeRelease(stm->refill_event);
SafeRelease(stm->input_available_event);
{
auto_lock lock(stm->stream_reset_lock);
close_wasapi_stream(stm);
}
// Need to call dtor to free the resource in owned_critical_section.
stm->stream_reset_lock.~owned_critical_section();
free(stm);
}
enum StreamDirection {
OUTPUT,
INPUT
};
int stream_start_one_side(cubeb_stream * stm, StreamDirection dir)
{
XASSERT((dir == OUTPUT && stm->output_client) ||
(dir == INPUT && stm->input_client));
HRESULT hr = dir == OUTPUT ? stm->output_client->Start() : stm->input_client->Start();
if (hr == AUDCLNT_E_DEVICE_INVALIDATED) {
LOG("audioclient invalidated for %s device, reconfiguring",
dir == OUTPUT ? "output" : "input");
BOOL ok = ResetEvent(stm->reconfigure_event);
if (!ok) {
LOG("resetting reconfig event failed for %s stream: %x",
dir == OUTPUT ? "output" : "input", GetLastError());
}
close_wasapi_stream(stm);
int r = setup_wasapi_stream(stm);
if (r != CUBEB_OK) {
LOG("reconfigure failed");
return r;
}
HRESULT hr2 = dir == OUTPUT ? stm->output_client->Start() : stm->input_client->Start();
if (FAILED(hr2)) {
LOG("could not start the %s stream after reconfig: %x",
dir == OUTPUT ? "output" : "input", hr);
return CUBEB_ERROR;
}
} else if (FAILED(hr)) {
LOG("could not start the %s stream: %x.",
dir == OUTPUT ? "output" : "input", hr);
return CUBEB_ERROR;
}
return CUBEB_OK;
}
int wasapi_stream_start(cubeb_stream * stm)
{
auto_lock lock(stm->stream_reset_lock);
XASSERT(stm && !stm->thread && !stm->shutdown_event);
XASSERT(stm->output_client || stm->input_client);
stm->emergency_bailout = new std::atomic<bool>(false);
if (stm->output_client) {
int rv = stream_start_one_side(stm, OUTPUT);
if (rv != CUBEB_OK) {
return rv;
}
}
if (stm->input_client) {
int rv = stream_start_one_side(stm, INPUT);
if (rv != CUBEB_OK) {
return rv;
}
}
stm->shutdown_event = CreateEvent(NULL, 0, 0, NULL);
if (!stm->shutdown_event) {
LOG("Can't create the shutdown event, error: %x", GetLastError());
return CUBEB_ERROR;
}
stm->thread = (HANDLE) _beginthreadex(NULL, 512 * 1024, wasapi_stream_render_loop, stm, STACK_SIZE_PARAM_IS_A_RESERVATION, NULL);
if (stm->thread == NULL) {
LOG("could not create WASAPI render thread.");
return CUBEB_ERROR;
}
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STARTED);
return CUBEB_OK;
}
int wasapi_stream_stop(cubeb_stream * stm)
{
XASSERT(stm);
HRESULT hr;
{
auto_lock lock(stm->stream_reset_lock);
if (stm->output_client) {
hr = stm->output_client->Stop();
if (FAILED(hr)) {
LOG("could not stop AudioClient (output)");
return CUBEB_ERROR;
}
}
if (stm->input_client) {
hr = stm->input_client->Stop();
if (FAILED(hr)) {
LOG("could not stop AudioClient (input)");
return CUBEB_ERROR;
}
}
stm->state_callback(stm, stm->user_ptr, CUBEB_STATE_STOPPED);
}
if (stop_and_join_render_thread(stm)) {
// This is null if we've given the pointer to the other thread
if (stm->emergency_bailout.load()) {
delete stm->emergency_bailout.load();
stm->emergency_bailout = nullptr;
}
}
return CUBEB_OK;
}
int wasapi_stream_get_position(cubeb_stream * stm, uint64_t * position)
{
XASSERT(stm && position);
auto_lock lock(stm->stream_reset_lock);
if (!has_output(stm)) {
return CUBEB_ERROR;
}
/* Calculate how far behind the current stream head the playback cursor is. */
uint64_t stream_delay = static_cast<uint64_t>(current_stream_delay(stm) * stm->output_stream_params.rate);
/* Calculate the logical stream head in frames at the stream sample rate. */
uint64_t max_pos = stm->total_frames_written +
static_cast<uint64_t>(round(stm->frames_written * stream_to_mix_samplerate_ratio(stm->output_stream_params, stm->output_mix_params)));
*position = max_pos;
if (stream_delay <= *position) {
*position -= stream_delay;
}
if (*position < stm->prev_position) {
*position = stm->prev_position;
}
stm->prev_position = *position;
return CUBEB_OK;
}
int wasapi_stream_get_latency(cubeb_stream * stm, uint32_t * latency)
{
XASSERT(stm && latency);
if (!has_output(stm)) {
return CUBEB_ERROR;
}
auto_lock lock(stm->stream_reset_lock);
/* The GetStreamLatency method only works if the
AudioClient has been initialized. */
if (!stm->output_client) {
return CUBEB_ERROR;
}
REFERENCE_TIME latency_hns;
HRESULT hr = stm->output_client->GetStreamLatency(&latency_hns);
if (FAILED(hr)) {
return CUBEB_ERROR;
}
*latency = hns_to_frames(stm, latency_hns);
return CUBEB_OK;
}
int wasapi_stream_set_volume(cubeb_stream * stm, float volume)
{
auto_lock lock(stm->stream_reset_lock);
if (!has_output(stm)) {
return CUBEB_ERROR;
}
if (stream_set_volume(stm, volume) != CUBEB_OK) {
return CUBEB_ERROR;
}
stm->volume = volume;
return CUBEB_OK;
}
static char *
wstr_to_utf8(LPCWSTR str)
{
char * ret = NULL;
int size;
size = ::WideCharToMultiByte(CP_UTF8, 0, str, -1, ret, 0, NULL, NULL);
if (size > 0) {
ret = static_cast<char *>(malloc(size));
::WideCharToMultiByte(CP_UTF8, 0, str, -1, ret, size, NULL, NULL);
}
return ret;
}
static std::unique_ptr<const wchar_t[]>
utf8_to_wstr(char* str)
{
std::unique_ptr<wchar_t[]> ret;
int size;
size = ::MultiByteToWideChar(CP_UTF8, 0, str, -1, nullptr, 0);
if (size > 0) {
ret.reset(new wchar_t[size]);
::MultiByteToWideChar(CP_UTF8, 0, str, -1, ret.get(), size);
}
return std::move(ret);
}
static IMMDevice *
wasapi_get_device_node(IMMDeviceEnumerator * enumerator, IMMDevice * dev)
{
IMMDevice * ret = NULL;
IDeviceTopology * devtopo = NULL;
IConnector * connector = NULL;
if (SUCCEEDED(dev->Activate(__uuidof(IDeviceTopology), CLSCTX_ALL, NULL, (void**)&devtopo)) &&
SUCCEEDED(devtopo->GetConnector(0, &connector))) {
LPWSTR filterid;
if (SUCCEEDED(connector->GetDeviceIdConnectedTo(&filterid))) {
if (FAILED(enumerator->GetDevice(filterid, &ret)))
ret = NULL;
CoTaskMemFree(filterid);
}
}
SafeRelease(connector);
SafeRelease(devtopo);
return ret;
}
static BOOL
wasapi_is_default_device(EDataFlow flow, ERole role, LPCWSTR device_id,
IMMDeviceEnumerator * enumerator)
{
BOOL ret = FALSE;
IMMDevice * dev;
HRESULT hr;
hr = enumerator->GetDefaultAudioEndpoint(flow, role, &dev);
if (SUCCEEDED(hr)) {
LPWSTR defdevid = NULL;
if (SUCCEEDED(dev->GetId(&defdevid)))
ret = (wcscmp(defdevid, device_id) == 0);
if (defdevid != NULL)
CoTaskMemFree(defdevid);
SafeRelease(dev);
}
return ret;
}
static cubeb_device_info *
wasapi_create_device(IMMDeviceEnumerator * enumerator, IMMDevice * dev)
{
IMMEndpoint * endpoint = NULL;
IMMDevice * devnode = NULL;
IAudioClient * client = NULL;
cubeb_device_info * ret = NULL;
EDataFlow flow;
LPWSTR device_id = NULL;
DWORD state = DEVICE_STATE_NOTPRESENT;
IPropertyStore * propstore = NULL;
PROPVARIANT propvar;
REFERENCE_TIME def_period, min_period;
HRESULT hr;
PropVariantInit(&propvar);
hr = dev->QueryInterface(IID_PPV_ARGS(&endpoint));
if (FAILED(hr)) goto done;
hr = endpoint->GetDataFlow(&flow);
if (FAILED(hr)) goto done;
hr = dev->GetId(&device_id);
if (FAILED(hr)) goto done;
hr = dev->OpenPropertyStore(STGM_READ, &propstore);
if (FAILED(hr)) goto done;
hr = dev->GetState(&state);
if (FAILED(hr)) goto done;
ret = (cubeb_device_info *)calloc(1, sizeof(cubeb_device_info));
ret->devid = ret->device_id = wstr_to_utf8(device_id);
hr = propstore->GetValue(PKEY_Device_FriendlyName, &propvar);
if (SUCCEEDED(hr))
ret->friendly_name = wstr_to_utf8(propvar.pwszVal);
devnode = wasapi_get_device_node(enumerator, dev);
if (devnode != NULL) {
IPropertyStore * ps = NULL;
hr = devnode->OpenPropertyStore(STGM_READ, &ps);
if (FAILED(hr)) goto done;
PropVariantClear(&propvar);
hr = ps->GetValue(PKEY_Device_InstanceId, &propvar);
if (SUCCEEDED(hr)) {
ret->group_id = wstr_to_utf8(propvar.pwszVal);
}
SafeRelease(ps);
}
ret->preferred = CUBEB_DEVICE_PREF_NONE;
if (wasapi_is_default_device(flow, eConsole, device_id, enumerator))
ret->preferred = (cubeb_device_pref)(ret->preferred | CUBEB_DEVICE_PREF_MULTIMEDIA);
if (wasapi_is_default_device(flow, eCommunications, device_id, enumerator))
ret->preferred = (cubeb_device_pref)(ret->preferred | CUBEB_DEVICE_PREF_VOICE);
if (wasapi_is_default_device(flow, eConsole, device_id, enumerator))
ret->preferred = (cubeb_device_pref)(ret->preferred | CUBEB_DEVICE_PREF_NOTIFICATION);
if (flow == eRender) ret->type = CUBEB_DEVICE_TYPE_OUTPUT;
else if (flow == eCapture) ret->type = CUBEB_DEVICE_TYPE_INPUT;
switch (state) {
case DEVICE_STATE_ACTIVE:
ret->state = CUBEB_DEVICE_STATE_ENABLED;
break;
case DEVICE_STATE_UNPLUGGED:
ret->state = CUBEB_DEVICE_STATE_UNPLUGGED;
break;
default:
ret->state = CUBEB_DEVICE_STATE_DISABLED;
break;
};
ret->format = CUBEB_DEVICE_FMT_F32NE; /* cubeb only supports 32bit float at the moment */
ret->default_format = CUBEB_DEVICE_FMT_F32NE;
PropVariantClear(&propvar);
hr = propstore->GetValue(PKEY_AudioEngine_DeviceFormat, &propvar);
if (SUCCEEDED(hr) && propvar.vt == VT_BLOB) {
if (propvar.blob.cbSize == sizeof(PCMWAVEFORMAT)) {
const PCMWAVEFORMAT * pcm = reinterpret_cast<const PCMWAVEFORMAT *>(propvar.blob.pBlobData);
ret->max_rate = ret->min_rate = ret->default_rate = pcm->wf.nSamplesPerSec;
ret->max_channels = pcm->wf.nChannels;
} else if (propvar.blob.cbSize >= sizeof(WAVEFORMATEX)) {
WAVEFORMATEX* wfx = reinterpret_cast<WAVEFORMATEX*>(propvar.blob.pBlobData);
if (propvar.blob.cbSize >= sizeof(WAVEFORMATEX) + wfx->cbSize ||
wfx->wFormatTag == WAVE_FORMAT_PCM) {
ret->max_rate = ret->min_rate = ret->default_rate = wfx->nSamplesPerSec;
ret->max_channels = wfx->nChannels;
}
}
}
if (SUCCEEDED(dev->Activate(__uuidof(IAudioClient), CLSCTX_INPROC_SERVER, NULL, (void**)&client)) &&
SUCCEEDED(client->GetDevicePeriod(&def_period, &min_period))) {
ret->latency_lo = hns_to_frames(ret->default_rate, min_period);
ret->latency_hi = hns_to_frames(ret->default_rate, def_period);
} else {
ret->latency_lo = 0;
ret->latency_hi = 0;
}
SafeRelease(client);
done:
SafeRelease(devnode);
SafeRelease(endpoint);
SafeRelease(propstore);
if (device_id != NULL)
CoTaskMemFree(device_id);
PropVariantClear(&propvar);
return ret;
}
static int
wasapi_enumerate_devices(cubeb * context, cubeb_device_type type,
cubeb_device_collection ** out)
{
auto_com com;
IMMDeviceEnumerator * enumerator;
IMMDeviceCollection * collection;
IMMDevice * dev;
cubeb_device_info * cur;
HRESULT hr;
UINT cc, i;
EDataFlow flow;
*out = NULL;
if (!com.ok())
return CUBEB_ERROR;
hr = CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL,
CLSCTX_INPROC_SERVER, IID_PPV_ARGS(&enumerator));
if (FAILED(hr)) {
LOG("Could not get device enumerator: %x", hr);
return CUBEB_ERROR;
}
if (type == CUBEB_DEVICE_TYPE_OUTPUT) flow = eRender;
else if (type == CUBEB_DEVICE_TYPE_INPUT) flow = eCapture;
else if (type & (CUBEB_DEVICE_TYPE_INPUT | CUBEB_DEVICE_TYPE_INPUT)) flow = eAll;
else return CUBEB_ERROR;
hr = enumerator->EnumAudioEndpoints(flow, DEVICE_STATEMASK_ALL, &collection);
if (FAILED(hr)) {
LOG("Could not enumerate audio endpoints: %x", hr);
return CUBEB_ERROR;
}
hr = collection->GetCount(&cc);
if (FAILED(hr)) {
LOG("IMMDeviceCollection::GetCount() failed: %x", hr);
return CUBEB_ERROR;
}
*out = (cubeb_device_collection *) malloc(sizeof(cubeb_device_collection) +
sizeof(cubeb_device_info*) * (cc > 0 ? cc - 1 : 0));
if (!*out) {
return CUBEB_ERROR;
}
(*out)->count = 0;
for (i = 0; i < cc; i++) {
hr = collection->Item(i, &dev);
if (FAILED(hr)) {
LOG("IMMDeviceCollection::Item(%u) failed: %x", i-1, hr);
} else if ((cur = wasapi_create_device(enumerator, dev)) != NULL) {
(*out)->device[(*out)->count++] = cur;
}
}
SafeRelease(collection);
SafeRelease(enumerator);
return CUBEB_OK;
}
cubeb_ops const wasapi_ops = {
/*.init =*/ wasapi_init,
/*.get_backend_id =*/ wasapi_get_backend_id,
/*.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,
/*.enumerate_devices =*/ wasapi_enumerate_devices,
/*.destroy =*/ wasapi_destroy,
/*.stream_init =*/ wasapi_stream_init,
/*.stream_destroy =*/ wasapi_stream_destroy,
/*.stream_start =*/ wasapi_stream_start,
/*.stream_stop =*/ wasapi_stream_stop,
/*.stream_get_position =*/ wasapi_stream_get_position,
/*.stream_get_latency =*/ wasapi_stream_get_latency,
/*.stream_set_volume =*/ wasapi_stream_set_volume,
/*.stream_set_panning =*/ NULL,
/*.stream_get_current_device =*/ NULL,
/*.stream_device_destroy =*/ NULL,
/*.stream_register_device_changed_callback =*/ NULL,
/*.register_device_collection_changed =*/ NULL
};
} // namespace anonymous
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