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-rw-r--r--audio/filter/af_scaletempo2.c33
-rw-r--r--audio/filter/af_scaletempo2_internals.c72
-rw-r--r--audio/filter/af_scaletempo2_internals.h1
-rw-r--r--audio/out/ao_coreaudio.c10
-rw-r--r--audio/out/ao_coreaudio_chmap.c2
-rw-r--r--audio/out/ao_coreaudio_chmap.h7
-rw-r--r--audio/out/ao_coreaudio_properties.h5
-rw-r--r--audio/out/ao_coreaudio_utils.c8
-rw-r--r--audio/out/ao_coreaudio_utils.h9
-rw-r--r--audio/out/ao_wasapi.c53
-rw-r--r--audio/out/ao_wasapi.h2
11 files changed, 81 insertions, 121 deletions
diff --git a/audio/filter/af_scaletempo2.c b/audio/filter/af_scaletempo2.c
index e43c29ac01..749e219454 100644
--- a/audio/filter/af_scaletempo2.c
+++ b/audio/filter/af_scaletempo2.c
@@ -8,7 +8,7 @@
#include "options/m_option.h"
struct priv {
- struct mp_scaletempo2 data;
+ struct mp_scaletempo2 *data;
struct mp_pin *in_pin;
struct mp_aframe *cur_format;
struct mp_aframe_pool *out_pool;
@@ -29,7 +29,7 @@ static void af_scaletempo2_process(struct mp_filter *f)
return;
while (!p->initialized || !p->pending ||
- !mp_scaletempo2_frames_available(&p->data, p->speed))
+ !mp_scaletempo2_frames_available(p->data, p->speed))
{
bool eof = false;
if (!p->pending || !mp_aframe_get_size(p->pending)) {
@@ -64,16 +64,16 @@ static void af_scaletempo2_process(struct mp_filter *f)
if (p->pending && !format_change && !p->sent_final) {
int frame_size = mp_aframe_get_size(p->pending);
uint8_t **planes = mp_aframe_get_data_ro(p->pending);
- int read = mp_scaletempo2_fill_input_buffer(&p->data,
+ int read = mp_scaletempo2_fill_input_buffer(p->data,
planes, frame_size, p->speed);
mp_aframe_skip_samples(p->pending, read);
}
if (final && p->pending && !p->sent_final) {
- mp_scaletempo2_set_final(&p->data);
+ mp_scaletempo2_set_final(p->data);
p->sent_final = true;
}
- if (mp_scaletempo2_frames_available(&p->data, p->speed)) {
+ if (mp_scaletempo2_frames_available(p->data, p->speed)) {
if (eof) {
mp_pin_out_repeat_eof(p->in_pin); // drain more next time
}
@@ -89,9 +89,9 @@ static void af_scaletempo2_process(struct mp_filter *f)
}
assert(p->pending);
- if (mp_scaletempo2_frames_available(&p->data, p->speed)) {
+ if (mp_scaletempo2_frames_available(p->data, p->speed)) {
struct mp_aframe *out = mp_aframe_new_ref(p->cur_format);
- int out_samples = p->data.ola_hop_size;
+ int out_samples = p->data->ola_hop_size;
if (mp_aframe_pool_allocate(p->out_pool, out, out_samples) < 0) {
talloc_free(out);
goto error;
@@ -101,14 +101,14 @@ static void af_scaletempo2_process(struct mp_filter *f)
uint8_t **planes = mp_aframe_get_data_rw(out);
assert(planes);
- assert(mp_aframe_get_planes(out) == p->data.channels);
+ assert(mp_aframe_get_planes(out) == p->data->channels);
- out_samples = mp_scaletempo2_fill_buffer(&p->data,
+ out_samples = mp_scaletempo2_fill_buffer(p->data,
(float**)planes, out_samples, p->speed);
double pts = mp_aframe_get_pts(p->pending);
if (pts != MP_NOPTS_VALUE) {
- double frame_delay = mp_scaletempo2_get_latency(&p->data, p->speed)
+ double frame_delay = mp_scaletempo2_get_latency(p->data, p->speed)
+ out_samples * p->speed;
mp_aframe_set_pts(out, pts - frame_delay / mp_aframe_get_effective_rate(out));
@@ -122,7 +122,7 @@ static void af_scaletempo2_process(struct mp_filter *f)
// reset the filter to ensure it stops generating audio
// and mp_scaletempo2_frames_available returns false
- mp_scaletempo2_reset(&p->data);
+ mp_scaletempo2_reset(p->data);
}
}
}
@@ -150,7 +150,7 @@ static bool init_scaletempo2(struct mp_filter *f)
p->sent_final = false;
mp_aframe_config_copy(p->cur_format, p->pending);
- mp_scaletempo2_init(&p->data, mp_aframe_get_channels(p->pending),
+ mp_scaletempo2_init(p->data, mp_aframe_get_channels(p->pending),
mp_aframe_get_rate(p->pending));
return true;
@@ -172,7 +172,7 @@ static bool af_scaletempo2_command(struct mp_filter *f, struct mp_filter_command
static void af_scaletempo2_reset(struct mp_filter *f)
{
struct priv *p = f->priv;
- mp_scaletempo2_reset(&p->data);
+ mp_scaletempo2_reset(p->data);
p->initialized = false;
TA_FREEP(&p->pending);
}
@@ -180,8 +180,8 @@ static void af_scaletempo2_reset(struct mp_filter *f)
static void af_scaletempo2_destroy(struct mp_filter *f)
{
struct priv *p = f->priv;
- mp_scaletempo2_destroy(&p->data);
- talloc_free(p->pending);
+ TA_FREEP(&p->data);
+ TA_FREEP(&p->pending);
}
static const struct mp_filter_info af_scaletempo2_filter = {
@@ -206,7 +206,8 @@ static struct mp_filter *af_scaletempo2_create(
mp_filter_add_pin(f, MP_PIN_OUT, "out");
struct priv *p = f->priv;
- p->data.opts = talloc_steal(p, options);
+ p->data = talloc_zero(p, struct mp_scaletempo2);
+ p->data->opts = talloc_steal(p, options);
p->speed = 1.0;
p->cur_format = talloc_steal(p, mp_aframe_create());
p->out_pool = mp_aframe_pool_create(p);
diff --git a/audio/filter/af_scaletempo2_internals.c b/audio/filter/af_scaletempo2_internals.c
index 924c0914b3..7f3a99638f 100644
--- a/audio/filter/af_scaletempo2_internals.c
+++ b/audio/filter/af_scaletempo2_internals.c
@@ -41,19 +41,15 @@ static bool in_interval(int n, struct interval q)
return n >= q.lo && n <= q.hi;
}
-static float **realloc_2d(float **p, int x, int y)
+static void alloc_sample_buffer(struct mp_scaletempo2 *p, float ***ptr, size_t size)
{
- float **array = realloc(p, sizeof(float*) * x + sizeof(float) * x * y);
- float* data = (float*) (array + x);
- for (int i = 0; i < x; ++i) {
- array[i] = data + i * y;
- }
- return array;
-}
+ talloc_free(*ptr);
-static void zero_2d(float **a, int x, int y)
-{
- memset(a + x, 0, sizeof(float) * x * y);
+ float **buff = talloc_array(p, float*, p->channels);
+ for (int i = 0; i < p->channels; ++i) {
+ buff[i] = talloc_array(buff, float, size);
+ }
+ *ptr = buff;
}
static void zero_2d_partial(float **a, int x, int y)
@@ -480,12 +476,6 @@ static bool can_perform_wsola(struct mp_scaletempo2 *p, double playback_rate)
return frames_needed(p, playback_rate) <= 0;
}
-static void resize_input_buffer(struct mp_scaletempo2 *p, int size)
-{
- p->input_buffer_size = size;
- p->input_buffer = realloc_2d(p->input_buffer, p->channels, size);
-}
-
// pad end with silence until a wsola iteration can be performed
static void add_input_buffer_final_silence(struct mp_scaletempo2 *p, double playback_rate)
{
@@ -493,11 +483,9 @@ static void add_input_buffer_final_silence(struct mp_scaletempo2 *p, double play
if (needed <= 0)
return; // no silence needed for iteration
- int required_size = needed + p->input_buffer_frames;
- if (required_size > p->input_buffer_size)
- resize_input_buffer(p, required_size);
-
+ int last_index = needed + p->input_buffer_frames - 1;
for (int i = 0; i < p->channels; ++i) {
+ MP_TARRAY_GROW(p, p->input_buffer[i], last_index);
float *ch_input = p->input_buffer[i];
for (int j = 0; j < needed; ++j) {
ch_input[p->input_buffer_frames + j] = 0.0f;
@@ -523,11 +511,9 @@ int mp_scaletempo2_fill_input_buffer(struct mp_scaletempo2 *p,
if (read == 0)
return 0;
- int required_size = read + p->input_buffer_frames;
- if (required_size > p->input_buffer_size)
- resize_input_buffer(p, required_size);
-
+ int last_index = read + p->input_buffer_frames - 1;
for (int i = 0; i < p->channels; ++i) {
+ MP_TARRAY_GROW(p, p->input_buffer[i], last_index);
memcpy(p->input_buffer[i] + p->input_buffer_frames,
planes[i], read * sizeof(float));
}
@@ -771,18 +757,6 @@ bool mp_scaletempo2_frames_available(struct mp_scaletempo2 *p, double playback_r
|| p->num_complete_frames > 0;
}
-void mp_scaletempo2_destroy(struct mp_scaletempo2 *p)
-{
- free(p->ola_window);
- free(p->transition_window);
- free(p->wsola_output);
- free(p->optimal_block);
- free(p->search_block);
- free(p->target_block);
- free(p->input_buffer);
- free(p->energy_candidate_blocks);
-}
-
void mp_scaletempo2_reset(struct mp_scaletempo2 *p)
{
p->input_buffer_frames = 0;
@@ -791,8 +765,6 @@ void mp_scaletempo2_reset(struct mp_scaletempo2 *p)
p->output_time = 0.0;
p->search_block_index = 0;
p->target_block_index = 0;
- // Clear the queue of decoded packets.
- zero_2d(p->wsola_output, p->channels, p->wsola_output_size);
p->num_complete_frames = 0;
p->wsola_output_started = false;
}
@@ -847,28 +819,26 @@ void mp_scaletempo2_init(struct mp_scaletempo2 *p, int channels, int rate)
// 1, ... |num_candidate_blocks|
p->search_block_center_offset = p->num_candidate_blocks / 2
+ (p->ola_window_size / 2 - 1);
- p->ola_window = realloc(p->ola_window, sizeof(float) * p->ola_window_size);
+ MP_RESIZE_ARRAY(p, p->ola_window, p->ola_window_size);
get_symmetric_hanning_window(p->ola_window_size, p->ola_window);
- p->transition_window = realloc(p->transition_window,
- sizeof(float) * p->ola_window_size * 2);
+ MP_RESIZE_ARRAY(p, p->transition_window, p->ola_window_size * 2);
get_symmetric_hanning_window(2 * p->ola_window_size, p->transition_window);
p->wsola_output_size = p->ola_window_size + p->ola_hop_size;
- p->wsola_output = realloc_2d(p->wsola_output, p->channels, p->wsola_output_size);
- // Initialize for overlap-and-add of the first block.
- zero_2d(p->wsola_output, p->channels, p->wsola_output_size);
+ alloc_sample_buffer(p, &p->wsola_output, p->wsola_output_size);
// Auxiliary containers.
- p->optimal_block = realloc_2d(p->optimal_block, p->channels, p->ola_window_size);
+ alloc_sample_buffer(p, &p->optimal_block, p->ola_window_size);
p->search_block_size = p->num_candidate_blocks + (p->ola_window_size - 1);
- p->search_block = realloc_2d(p->search_block, p->channels, p->search_block_size);
- p->target_block = realloc_2d(p->target_block, p->channels, p->ola_window_size);
+ alloc_sample_buffer(p, &p->search_block, p->search_block_size);
+ alloc_sample_buffer(p, &p->target_block, p->ola_window_size);
- resize_input_buffer(p, 4 * MPMAX(p->ola_window_size, p->search_block_size));
p->input_buffer_frames = 0;
p->input_buffer_final_frames = 0;
p->input_buffer_added_silence = 0;
+ size_t initial_size = 4 * MPMAX(p->ola_window_size, p->search_block_size);
+ alloc_sample_buffer(p, &p->input_buffer, initial_size);
- p->energy_candidate_blocks = realloc(p->energy_candidate_blocks,
- sizeof(float) * p->channels * p->num_candidate_blocks);
+ MP_RESIZE_ARRAY(p, p->energy_candidate_blocks,
+ p->channels * p->num_candidate_blocks);
}
diff --git a/audio/filter/af_scaletempo2_internals.h b/audio/filter/af_scaletempo2_internals.h
index 6c3c94c0a9..254d0a7704 100644
--- a/audio/filter/af_scaletempo2_internals.h
+++ b/audio/filter/af_scaletempo2_internals.h
@@ -110,7 +110,6 @@ struct mp_scaletempo2 {
float **target_block;
// Buffered audio data.
float **input_buffer;
- int input_buffer_size;
int input_buffer_frames;
// How many frames in |input_buffer| need to be flushed by padding with
// silence to process the final packet. While this is nonzero, the filter
diff --git a/audio/out/ao_coreaudio.c b/audio/out/ao_coreaudio.c
index ae743c90b2..4abfac38ee 100644
--- a/audio/out/ao_coreaudio.c
+++ b/audio/out/ao_coreaudio.c
@@ -89,14 +89,8 @@ static OSStatus render_cb_lpcm(void *ctx, AudioUnitRenderActionFlags *aflags,
int64_t end = mp_time_ns();
end += p->hw_latency_ns + ca_get_latency(ts) + ca_frames_to_ns(ao, frames);
- int samples = ao_read_data(ao, planes, frames, end);
-
- if (samples == 0)
- *aflags |= kAudioUnitRenderAction_OutputIsSilence;
-
- for (int n = 0; n < buffer_list->mNumberBuffers; n++)
- buffer_list->mBuffers[n].mDataByteSize = samples * ao->sstride;
-
+ // don't use the returned sample count since CoreAudio always expects full frames
+ ao_read_data(ao, planes, frames, end);
return noErr;
}
diff --git a/audio/out/ao_coreaudio_chmap.c b/audio/out/ao_coreaudio_chmap.c
index 5a129c4e67..b45cf7799c 100644
--- a/audio/out/ao_coreaudio_chmap.c
+++ b/audio/out/ao_coreaudio_chmap.c
@@ -195,7 +195,7 @@ AudioChannelLabel mp_speaker_id_to_ca_label(int speaker_id)
return -1; // kAudioChannelLabel_Unknown
}
-#if HAVE_COREAUDIO
+#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
void ca_log_layout(struct ao *ao, int l, AudioChannelLayout *layout)
{
if (!mp_msg_test(ao->log, l))
diff --git a/audio/out/ao_coreaudio_chmap.h b/audio/out/ao_coreaudio_chmap.h
index 0b21e8330c..5286f98e63 100644
--- a/audio/out/ao_coreaudio_chmap.h
+++ b/audio/out/ao_coreaudio_chmap.h
@@ -21,17 +21,12 @@
#include "config.h"
#include <AudioToolbox/AudioToolbox.h>
-#if HAVE_AVFOUNDATION || HAVE_AUDIOUNIT
-#undef HAVE_COREAUDIO
-#define HAVE_COREAUDIO 1
-#endif
-
struct mp_chmap;
int ca_label_to_mp_speaker_id(AudioChannelLabel label);
AudioChannelLabel mp_speaker_id_to_ca_label(int speaker_id);
-#if HAVE_COREAUDIO
+#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
AudioChannelLayout *ca_find_standard_layout(void *talloc_ctx, AudioChannelLayout *l);
void ca_log_layout(struct ao *ao, int l, AudioChannelLayout *layout);
bool ca_init_chmap(struct ao *ao, AudioDeviceID device);
diff --git a/audio/out/ao_coreaudio_properties.h b/audio/out/ao_coreaudio_properties.h
index 2c9c5657bb..f29396883d 100644
--- a/audio/out/ao_coreaudio_properties.h
+++ b/audio/out/ao_coreaudio_properties.h
@@ -23,11 +23,6 @@
#include "internal.h"
-#if HAVE_AVFOUNDATION || HAVE_AUDIOUNIT
-#undef HAVE_COREAUDIO
-#define HAVE_COREAUDIO 1
-#endif
-
// CoreAudio names are way too verbose
#define ca_sel AudioObjectPropertySelector
#define ca_scope AudioObjectPropertyScope
diff --git a/audio/out/ao_coreaudio_utils.c b/audio/out/ao_coreaudio_utils.c
index e74092a8d2..44604e6df2 100644
--- a/audio/out/ao_coreaudio_utils.c
+++ b/audio/out/ao_coreaudio_utils.c
@@ -28,14 +28,14 @@
#include "osdep/semaphore.h"
#include "audio/format.h"
-#if HAVE_COREAUDIO
+#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
#include "audio/out/ao_coreaudio_properties.h"
#include <CoreAudio/HostTime.h>
#else
#include <mach/mach_time.h>
#endif
-#if HAVE_COREAUDIO
+#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
static bool ca_is_output_device(struct ao *ao, AudioDeviceID dev)
{
size_t n_buffers;
@@ -300,7 +300,7 @@ int64_t ca_frames_to_ns(struct ao *ao, uint32_t frames)
int64_t ca_get_latency(const AudioTimeStamp *ts)
{
-#if HAVE_COREAUDIO
+#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
uint64_t out = AudioConvertHostTimeToNanos(ts->mHostTime);
uint64_t now = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
@@ -323,7 +323,7 @@ int64_t ca_get_latency(const AudioTimeStamp *ts)
#endif
}
-#if HAVE_COREAUDIO
+#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
bool ca_stream_supports_compressed(struct ao *ao, AudioStreamID stream)
{
AudioStreamRangedDescription *formats = NULL;
diff --git a/audio/out/ao_coreaudio_utils.h b/audio/out/ao_coreaudio_utils.h
index 699ffde9a7..d11506c059 100644
--- a/audio/out/ao_coreaudio_utils.h
+++ b/audio/out/ao_coreaudio_utils.h
@@ -29,11 +29,6 @@
#include "internal.h"
#include "osdep/utils-mac.h"
-#if HAVE_AVFOUNDATION || HAVE_AUDIOUNIT
-#undef HAVE_COREAUDIO
-#define HAVE_COREAUDIO 1
-#endif
-
bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message);
#define CHECK_CA_ERROR_L(label, message) \
@@ -52,7 +47,7 @@ bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message);
} while (0)
void ca_get_device_list(struct ao *ao, struct ao_device_list *list);
-#if HAVE_COREAUDIO
+#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device);
#endif
@@ -70,7 +65,7 @@ bool ca_asbd_is_better(AudioStreamBasicDescription *req,
int64_t ca_frames_to_ns(struct ao *ao, uint32_t frames);
int64_t ca_get_latency(const AudioTimeStamp *ts);
-#if HAVE_COREAUDIO
+#if HAVE_COREAUDIO || HAVE_AVFOUNDATION
bool ca_stream_supports_compressed(struct ao *ao, AudioStreamID stream);
OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid);
OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid);
diff --git a/audio/out/ao_wasapi.c b/audio/out/ao_wasapi.c
index 58d2acd09a..34c3b6193f 100644
--- a/audio/out/ao_wasapi.c
+++ b/audio/out/ao_wasapi.c
@@ -193,17 +193,19 @@ static void thread_resume(struct ao *ao)
thread_unpause(ao);
}
-static void set_state_and_wakeup_thread(struct ao *ao,
- enum wasapi_thread_state thread_state)
+static void thread_wakeup(void *ptr)
{
+ struct ao *ao = ptr;
struct wasapi_state *state = ao->priv;
- atomic_store(&state->thread_state, thread_state);
- SetEvent(state->hWake);
+ SetEvent(state->hUserWake);
}
-static void thread_process_dispatch(void *ptr)
+static void set_thread_state(struct ao *ao,
+ enum wasapi_thread_state thread_state)
{
- set_state_and_wakeup_thread(ptr, WASAPI_THREAD_DISPATCH);
+ struct wasapi_state *state = ao->priv;
+ atomic_store(&state->thread_state, thread_state);
+ thread_wakeup(ao);
}
static DWORD __stdcall AudioThread(void *lpParameter)
@@ -220,18 +222,25 @@ static DWORD __stdcall AudioThread(void *lpParameter)
MP_DBG(ao, "Entering dispatch loop\n");
while (true) {
- if (WaitForSingleObject(state->hWake, INFINITE) != WAIT_OBJECT_0)
- MP_ERR(ao, "Unexpected return value from WaitForSingleObject\n");
-
- int thread_state = atomic_load(&state->thread_state);
- switch (thread_state) {
- case WASAPI_THREAD_FEED:
+ HANDLE handles[] = {state->hWake, state->hUserWake};
+ switch (WaitForMultipleObjects(MP_ARRAY_SIZE(handles), handles, FALSE, INFINITE)) {
+ case WAIT_OBJECT_0:
// fill twice on under-full buffer (see comment in thread_feed)
if (thread_feed(ao) && thread_feed(ao))
MP_ERR(ao, "Unable to fill buffer fast enough\n");
+ continue;
+ case WAIT_OBJECT_0 + 1:
break;
- case WASAPI_THREAD_DISPATCH:
- mp_dispatch_queue_process(state->dispatch, 0);
+ default:
+ MP_ERR(ao, "Unexpected return value from WaitForMultipleObjects\n");
+ break;
+ }
+
+ mp_dispatch_queue_process(state->dispatch, 0);
+
+ int thread_state = atomic_load(&state->thread_state);
+ switch (thread_state) {
+ case WASAPI_THREAD_FEED:
break;
case WASAPI_THREAD_RESET:
thread_reset(ao);
@@ -267,8 +276,8 @@ static void uninit(struct ao *ao)
{
MP_DBG(ao, "Uninit wasapi\n");
struct wasapi_state *state = ao->priv;
- if (state->hWake)
- set_state_and_wakeup_thread(ao, WASAPI_THREAD_SHUTDOWN);
+ if (state->hWake && state->hUserWake)
+ set_thread_state(ao, WASAPI_THREAD_SHUTDOWN);
if (state->hAudioThread &&
WaitForSingleObject(state->hAudioThread, INFINITE) != WAIT_OBJECT_0)
@@ -279,6 +288,7 @@ static void uninit(struct ao *ao)
SAFE_DESTROY(state->hInitDone, CloseHandle(state->hInitDone));
SAFE_DESTROY(state->hWake, CloseHandle(state->hWake));
+ SAFE_DESTROY(state->hUserWake, CloseHandle(state->hUserWake));
SAFE_DESTROY(state->hAudioThread,CloseHandle(state->hAudioThread));
wasapi_change_uninit(ao);
@@ -312,14 +322,15 @@ static int init(struct ao *ao)
state->hInitDone = CreateEventW(NULL, FALSE, FALSE, NULL);
state->hWake = CreateEventW(NULL, FALSE, FALSE, NULL);
- if (!state->hInitDone || !state->hWake) {
+ state->hUserWake = CreateEventW(NULL, FALSE, FALSE, NULL);
+ if (!state->hInitDone || !state->hWake || !state->hUserWake) {
MP_FATAL(ao, "Error creating events\n");
uninit(ao);
return -1;
}
state->dispatch = mp_dispatch_create(state);
- mp_dispatch_set_wakeup_fn(state->dispatch, thread_process_dispatch, ao);
+ mp_dispatch_set_wakeup_fn(state->dispatch, thread_wakeup, ao);
state->init_ok = false;
state->hAudioThread = CreateThread(NULL, 0, &AudioThread, ao, 0, NULL);
@@ -474,17 +485,17 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
static void audio_reset(struct ao *ao)
{
- set_state_and_wakeup_thread(ao, WASAPI_THREAD_RESET);
+ set_thread_state(ao, WASAPI_THREAD_RESET);
}
static void audio_resume(struct ao *ao)
{
- set_state_and_wakeup_thread(ao, WASAPI_THREAD_RESUME);
+ set_thread_state(ao, WASAPI_THREAD_RESUME);
}
static bool audio_set_pause(struct ao *ao, bool paused)
{
- set_state_and_wakeup_thread(ao, paused ? WASAPI_THREAD_PAUSE : WASAPI_THREAD_UNPAUSE);
+ set_thread_state(ao, paused ? WASAPI_THREAD_PAUSE : WASAPI_THREAD_UNPAUSE);
return true;
}
diff --git a/audio/out/ao_wasapi.h b/audio/out/ao_wasapi.h
index e294e317c3..cbd4a3f933 100644
--- a/audio/out/ao_wasapi.h
+++ b/audio/out/ao_wasapi.h
@@ -48,7 +48,6 @@ void wasapi_change_uninit(struct ao* ao);
enum wasapi_thread_state {
WASAPI_THREAD_FEED = 0,
- WASAPI_THREAD_DISPATCH,
WASAPI_THREAD_RESUME,
WASAPI_THREAD_RESET,
WASAPI_THREAD_SHUTDOWN,
@@ -66,6 +65,7 @@ typedef struct wasapi_state {
HANDLE hWake; // thread wakeup event
atomic_int thread_state; // enum wasapi_thread_state (what to do on wakeup)
struct mp_dispatch_queue *dispatch; // for volume/mute/session display
+ HANDLE hUserWake; // mpv-requested wakeup event
// for setting the audio thread priority
HANDLE hTask;
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-02 21:09:36 +0000