summaryrefslogtreecommitdiffstats
path: root/audio
diff options
context:
space:
mode:
Diffstat (limited to 'audio')
-rw-r--r--audio/filter/af_scaletempo.c4
-rw-r--r--audio/out/ao.c19
-rw-r--r--audio/out/ao.h4
-rw-r--r--audio/out/ao_coreaudio.c65
-rw-r--r--audio/out/ao_coreaudio_properties.c6
-rw-r--r--audio/out/ao_coreaudio_properties.h2
-rw-r--r--audio/out/ao_coreaudio_utils.c110
-rw-r--r--audio/out/ao_coreaudio_utils.h31
-rw-r--r--audio/out/ao_wasapi.c12
9 files changed, 133 insertions, 120 deletions
diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c
index b12954f1af..d409ed85cf 100644
--- a/audio/filter/af_scaletempo.c
+++ b/audio/filter/af_scaletempo.c
@@ -444,11 +444,11 @@ static int control(struct af_instance* af, int cmd, void* arg)
if (s->speed_pitch) {
break;
}
- s->speed = *(float*)arg;
+ s->speed = *(double *)arg;
s->scale = s->speed * s->scale_nominal;
} else {
if (s->speed_pitch) {
- s->speed = 1 / *(float*)arg;
+ s->speed = 1 / *(double *)arg;
s->scale = s->speed * s->scale_nominal;
break;
}
diff --git a/audio/out/ao.c b/audio/out/ao.c
index 04e6fcda12..49fffb657d 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -30,6 +30,7 @@
#include "core/options.h"
#include "core/m_config.h"
#include "core/mp_msg.h"
+#include "core/mpv_global.h"
// there are some globals:
struct ao *global_ao;
@@ -121,27 +122,31 @@ const struct m_obj_list ao_obj_list = {
.allow_trailer = true,
};
-static struct ao *ao_create(bool probing, struct MPOpts *opts,
+static struct ao *ao_create(bool probing, struct mpv_global *global,
struct input_ctx *input_ctx,
struct encode_lavc_context *encode_lavc_ctx,
int samplerate, int format, struct mp_chmap channels,
char *name, char **args)
{
+ struct mp_log *log = mp_log_new(NULL, global->log, "ao");
struct m_obj_desc desc;
if (!m_obj_list_find(&desc, &ao_obj_list, bstr0(name))) {
- mp_tmsg(MSGT_CPLAYER, MSGL_ERR, "Audio output %s not found!\n", name);
+ mp_tmsg_log(log, MSGL_ERR, "Audio output %s not found!\n", name);
+ talloc_free(log);
return NULL;
};
struct ao *ao = talloc_ptrtype(NULL, ao);
+ talloc_steal(ao, log);
*ao = (struct ao) {
.driver = desc.p,
.probing = probing,
- .opts = opts,
+ .opts = global->opts,
.encode_lavc_ctx = encode_lavc_ctx,
.input_ctx = input_ctx,
.samplerate = samplerate,
.channels = channels,
.format = format,
+ .log = mp_log_new(ao, log, name),
};
if (ao->driver->encode != !!ao->encode_lavc_ctx)
goto error;
@@ -158,19 +163,19 @@ error:
return NULL;
}
-struct ao *ao_init_best(struct MPOpts *opts,
+struct ao *ao_init_best(struct mpv_global *global,
struct input_ctx *input_ctx,
struct encode_lavc_context *encode_lavc_ctx,
int samplerate, int format, struct mp_chmap channels)
{
- struct m_obj_settings *ao_list = opts->audio_driver_list;
+ struct m_obj_settings *ao_list = global->opts->audio_driver_list;
if (ao_list && ao_list[0].name) {
for (int n = 0; ao_list[n].name; n++) {
if (strlen(ao_list[n].name) == 0)
goto autoprobe;
mp_tmsg(MSGT_AO, MSGL_V, "Trying preferred audio driver '%s'\n",
ao_list[n].name);
- struct ao *ao = ao_create(false, opts, input_ctx, encode_lavc_ctx,
+ struct ao *ao = ao_create(false, global, input_ctx, encode_lavc_ctx,
samplerate, format, channels,
ao_list[n].name, ao_list[n].attribs);
if (ao)
@@ -183,7 +188,7 @@ struct ao *ao_init_best(struct MPOpts *opts,
autoprobe:
// now try the rest...
for (int i = 0; audio_out_drivers[i]; i++) {
- struct ao *ao = ao_create(true, opts, input_ctx, encode_lavc_ctx,
+ struct ao *ao = ao_create(true, global, input_ctx, encode_lavc_ctx,
samplerate, format, channels,
(char *)audio_out_drivers[i]->info->short_name, NULL);
if (ao)
diff --git a/audio/out/ao.h b/audio/out/ao.h
index 25cd12a0f1..65e5a3fc6b 100644
--- a/audio/out/ao.h
+++ b/audio/out/ao.h
@@ -93,11 +93,13 @@ struct ao {
struct encode_lavc_context *encode_lavc_ctx;
struct MPOpts *opts;
struct input_ctx *input_ctx;
+ struct mp_log *log; // Using e.g. "[ao/coreaudio]" as prefix
};
extern char *ao_subdevice;
-struct ao *ao_init_best(struct MPOpts *opts,
+struct mpv_global;
+struct ao *ao_init_best(struct mpv_global *global,
struct input_ctx *input_ctx,
struct encode_lavc_context *encode_lavc_ctx,
int samplerate, int format, struct mp_chmap channels);
diff --git a/audio/out/ao_coreaudio.c b/audio/out/ao_coreaudio.c
index 718c71187d..59e1f82f36 100644
--- a/audio/out/ao_coreaudio.c
+++ b/audio/out/ao_coreaudio.c
@@ -46,10 +46,10 @@ static void audio_pause(struct ao *ao);
static void audio_resume(struct ao *ao);
static void reset(struct ao *ao);
-static void print_buffer(struct mp_ring *buffer)
+static void print_buffer(struct ao *ao, struct mp_ring *buffer)
{
void *tctx = talloc_new(NULL);
- ca_msg(MSGL_V, "%s\n", mp_ring_repr(buffer, tctx));
+ MP_VERBOSE(ao, "%s\n", mp_ring_repr(buffer, tctx));
talloc_free(tctx);
}
@@ -109,7 +109,12 @@ static OSStatus render_cb_lpcm(void *ctx, AudioUnitRenderActionFlags *aflags,
AudioBuffer buf = buffer_list->mBuffers[0];
int requested = buf.mDataByteSize;
- buf.mDataByteSize = mp_ring_read(p->buffer, buf.mData, requested);
+ if (mp_ring_buffered(p->buffer) < requested) {
+ MP_VERBOSE(ao, "buffer underrun\n");
+ audio_pause(ao);
+ } else {
+ mp_ring_read(p->buffer, buf.mData, requested);
+ }
return noErr;
}
@@ -192,7 +197,7 @@ coreaudio_error:
return CONTROL_ERROR;
}
-static void print_list(void)
+static void print_list(struct ao *ao)
{
char *help = talloc_strdup(NULL, "Available output devices:\n");
@@ -221,7 +226,7 @@ static void print_list(void)
talloc_free(devs);
coreaudio_error:
- ca_msg(MSGL_INFO, "%s", help);
+ MP_INFO(ao, "%s", help);
talloc_free(help);
}
@@ -233,7 +238,7 @@ static int init(struct ao *ao)
OSStatus err;
struct priv *p = ao->priv;
- if (p->opt_list) print_list();
+ if (p->opt_list) print_list(ao);
struct priv_d *d = talloc_zero(p, struct priv_d);
@@ -266,9 +271,8 @@ static int init(struct ao *ao)
err = CA_GET_STR(selected_device, kAudioObjectPropertyName, &device_name);
CHECK_CA_ERROR("could not get selected audio device name");
- ca_msg(MSGL_V,
- "selected audio output device: %s (%" PRIu32 ")\n",
- device_name, selected_device);
+ MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
+ device_name, selected_device);
talloc_free(device_name);
@@ -278,7 +282,7 @@ static int init(struct ao *ao)
bool supports_digital = false;
/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
if (AF_FORMAT_IS_AC3(ao->format)) {
- if (ca_device_supports_digital(selected_device))
+ if (ca_device_supports_digital(ao, selected_device))
supports_digital = true;
}
@@ -293,7 +297,7 @@ static int init(struct ao *ao)
uint32_t *bitmaps;
size_t n_bitmaps;
- ca_bitmaps_from_layouts(layouts, n_layouts, &bitmaps, &n_bitmaps);
+ ca_bitmaps_from_layouts(ao, layouts, n_layouts, &bitmaps, &n_bitmaps);
talloc_free(layouts);
struct mp_chmap_sel chmap_sel = {0};
@@ -339,7 +343,7 @@ static int init(struct ao *ao)
asbd.mFramesPerPacket * asbd.mChannelsPerFrame *
(asbd.mBitsPerChannel / 8);
- ca_print_asbd("source format:", &asbd);
+ ca_print_asbd(ao, "source format:", &asbd);
if (supports_digital)
return init_digital(ao, asbd);
@@ -358,7 +362,9 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
AudioComponentDescription desc = (AudioComponentDescription) {
.componentType = kAudioUnitType_Output,
- .componentSubType = kAudioUnitSubType_HALOutput,
+ .componentSubType = (p->opt_device_id < 0) ?
+ kAudioUnitSubType_DefaultOutput :
+ kAudioUnitSubType_HALOutput,
.componentManufacturer = kAudioUnitManufacturer_Apple,
.componentFlags = 0,
.componentFlagsMask = 0,
@@ -366,7 +372,7 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
AudioComponent comp = AudioComponentFindNext(NULL, &desc);
if (comp == NULL) {
- ca_msg(MSGL_ERR, "unable to find audio component\n");
+ MP_ERR(ao, "unable to find audio component\n");
goto coreaudio_error;
}
@@ -410,7 +416,7 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
}
p->buffer = mp_ring_new(p, get_ring_size(ao));
- print_buffer(p->buffer);
+ print_buffer(ao, p->buffer);
AURenderCallbackStruct render_cb = (AURenderCallbackStruct) {
.inputProc = render_cb_lpcm,
@@ -447,14 +453,14 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
CHECK_CA_WARN("could not check whether device is alive");
if (!is_alive)
- ca_msg(MSGL_WARN, "device is not alive\n");
+ MP_WARN(ao , "device is not alive\n");
p->is_digital = 1;
err = ca_lock_device(p->device, &d->hog_pid);
CHECK_CA_WARN("failed to set hogmode");
- err = ca_disable_mixing(p->device, &d->changed_mixing);
+ err = ca_disable_mixing(ao, p->device, &d->changed_mixing);
CHECK_CA_WARN("failed to disable mixing");
AudioStreamID *streams;
@@ -467,7 +473,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
CHECK_CA_ERROR("could not get number of streams");
for (int i = 0; i < n_streams && d->stream_idx < 0; i++) {
- bool digital = ca_stream_supports_digital(streams[i]);
+ bool digital = ca_stream_supports_digital(ao, streams[i]);
if (digital) {
err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat,
@@ -518,11 +524,11 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
talloc_free(streams);
if (d->stream_idx < 0) {
- ca_msg(MSGL_WARN, "can't find any digital output stream format\n");
+ MP_WARN(ao , "can't find any digital output stream format\n");
goto coreaudio_error;
}
- if (!ca_change_format(d->stream, d->stream_asbd))
+ if (!ca_change_format(ao, d->stream, d->stream_asbd))
goto coreaudio_error;
void *changed = (void *) &(d->stream_asbd_changed);
@@ -537,8 +543,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
ao->format = AF_FORMAT_AC3_LE;
else if (d->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
- ca_msg(MSGL_WARN,
- "stream has non-native byte order, digital output may fail\n");
+ MP_WARN(ao, "stream has non-native byte order, output may fail\n");
ao->samplerate = d->stream_asbd.mSampleRate;
ao->bps = ao->samplerate *
@@ -546,7 +551,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
d->stream_asbd.mFramesPerPacket);
p->buffer = mp_ring_new(p, get_ring_size(ao));
- print_buffer(p->buffer);
+ print_buffer(ao, p->buffer);
err = AudioDeviceCreateIOProcID(p->device,
(AudioDeviceIOProc)render_cb_digital,
@@ -573,11 +578,11 @@ static int play(struct ao *ao, void *output_samples, int num_bytes, int flags)
// Check whether we need to reset the digital output stream.
if (p->is_digital && d->stream_asbd_changed) {
d->stream_asbd_changed = 0;
- if (ca_stream_supports_digital(d->stream)) {
- if (!ca_change_format(d->stream, d->stream_asbd)) {
- ca_msg(MSGL_WARN, "can't restore digital output\n");
+ if (ca_stream_supports_digital(ao, d->stream)) {
+ if (!ca_change_format(ao, d->stream, d->stream_asbd)) {
+ MP_WARN(ao , "can't restore digital output\n");
} else {
- ca_msg(MSGL_WARN, "restoring digital output succeeded.\n");
+ MP_WARN(ao, "restoring digital output succeeded.\n");
reset(ao);
}
}
@@ -635,10 +640,10 @@ static void uninit(struct ao *ao, bool immed)
err = AudioDeviceDestroyIOProcID(p->device, d->render_cb);
CHECK_CA_WARN("failed to remove device render callback");
- if (!ca_change_format(d->stream, d->original_asbd))
- ca_msg(MSGL_WARN, "can't revert to original device format");
+ if (!ca_change_format(ao, d->stream, d->original_asbd))
+ MP_WARN(ao, "can't revert to original device format");
- err = ca_enable_mixing(p->device, d->changed_mixing);
+ err = ca_enable_mixing(ao, p->device, d->changed_mixing);
CHECK_CA_WARN("can't re-enable mixing");
err = ca_unlock_device(p->device, &d->hog_pid);
diff --git a/audio/out/ao_coreaudio_properties.c b/audio/out/ao_coreaudio_properties.c
index 300f27a8f0..67287e7427 100644
--- a/audio/out/ao_coreaudio_properties.c
+++ b/audio/out/ao_coreaudio_properties.c
@@ -60,13 +60,13 @@ OSStatus ca_get_ary(AudioObjectID id, ca_scope scope, ca_sel selector,
};
err = AudioObjectGetPropertyDataSize(id, &p_addr, 0, NULL, &p_size);
- CHECK_CA_ERROR("can't fetch property size");
+ CHECK_CA_ERROR_SILENT_L(coreaudio_error);
*data = talloc_size(NULL, p_size);
*elements = p_size / element_size;
err = ca_get(id, scope, selector, p_size, *data);
- CHECK_CA_ERROR_L(coreaudio_error_free, "can't fetch property data");
+ CHECK_CA_ERROR_SILENT_L(coreaudio_error_free);
return err;
coreaudio_error_free:
@@ -81,7 +81,7 @@ OSStatus ca_get_str(AudioObjectID id, ca_scope scope, ca_sel selector,
CFStringRef string;
OSStatus err =
ca_get(id, scope, selector, sizeof(CFStringRef), (void **)&string);
- CHECK_CA_ERROR("Can't fetch string property");
+ CHECK_CA_ERROR_SILENT_L(coreaudio_error);
CFIndex size =
CFStringGetMaximumSizeForEncoding(
diff --git a/audio/out/ao_coreaudio_properties.h b/audio/out/ao_coreaudio_properties.h
index 88ba889c29..ee7f83e6b7 100644
--- a/audio/out/ao_coreaudio_properties.h
+++ b/audio/out/ao_coreaudio_properties.h
@@ -25,7 +25,7 @@
#define ca_sel AudioObjectPropertySelector
#define ca_scope AudioObjectPropertyScope
#define CA_GLOBAL kAudioObjectPropertyScopeGlobal
-#define CA_OUTPUT kAudioObjectPropertyScopeOutput
+#define CA_OUTPUT kAudioDevicePropertyScopeOutput
OSStatus ca_get(AudioObjectID id, ca_scope scope, ca_sel selector,
uint32_t size, void *data);
diff --git a/audio/out/ao_coreaudio_utils.c b/audio/out/ao_coreaudio_utils.c
index 68c87f9cc6..2c12b562cf 100644
--- a/audio/out/ao_coreaudio_utils.c
+++ b/audio/out/ao_coreaudio_utils.c
@@ -26,9 +26,6 @@
#include "audio/out/ao_coreaudio_properties.h"
#include "osdep/timer.h"
-#define ca_msg(a, b ...) mp_msg(MSGT_AO, a, "AO: [coreaudio] " b)
-#define CA_CFSTR_ENCODING kCFStringEncodingASCII
-
char *fourcc_repr(void *talloc_ctx, uint32_t code)
{
// Extract FourCC letters from the uint32_t and finde out if it's a valid
@@ -55,24 +52,24 @@ char *fourcc_repr(void *talloc_ctx, uint32_t code)
return repr;
}
-bool check_ca_st(int level, OSStatus code, const char *message)
+bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message)
{
if (code == noErr) return true;
char *error_string = fourcc_repr(NULL, code);
- ca_msg(level, "%s (%s)\n", message, error_string);
+ mp_msg_log(ao->log, level, "%s (%s)\n", message, error_string);
talloc_free(error_string);
return false;
}
-void ca_print_asbd(const char *description,
+void ca_print_asbd(struct ao *ao, const char *description,
const AudioStreamBasicDescription *asbd)
{
uint32_t flags = asbd->mFormatFlags;
char *format = fourcc_repr(NULL, asbd->mFormatID);
- ca_msg(MSGL_V,
+ MP_VERBOSE(ao,
"%s %7.1fHz %" PRIu32 "bit [%s]"
"[%" PRIu32 "][%" PRIu32 "][%" PRIu32 "]"
"[%" PRIu32 "][%" PRIu32 "] "
@@ -101,7 +98,7 @@ bool ca_format_is_digital(AudioStreamBasicDescription asbd)
return false;
}
-bool ca_stream_supports_digital(AudioStreamID stream)
+bool ca_stream_supports_digital(struct ao *ao, AudioStreamID stream)
{
AudioStreamRangedDescription *formats = NULL;
size_t n_formats;
@@ -114,7 +111,7 @@ bool ca_stream_supports_digital(AudioStreamID stream)
for (int i = 0; i < n_formats; i++) {
AudioStreamBasicDescription asbd = formats[i].mFormat;
- ca_print_asbd("supported format:", &(asbd));
+ ca_print_asbd(ao, "supported format:", &(asbd));
if (ca_format_is_digital(asbd)) {
talloc_free(formats);
return true;
@@ -126,7 +123,7 @@ coreaudio_error:
return false;
}
-bool ca_device_supports_digital(AudioDeviceID device)
+bool ca_device_supports_digital(struct ao *ao, AudioDeviceID device)
{
AudioStreamID *streams = NULL;
size_t n_streams;
@@ -138,7 +135,7 @@ bool ca_device_supports_digital(AudioDeviceID device)
CHECK_CA_ERROR("could not get number of streams.");
for (int i = 0; i < n_streams; i++) {
- if (ca_stream_supports_digital(streams[i])) {
+ if (ca_stream_supports_digital(ao, streams[i])) {
talloc_free(streams);
return true;
}
@@ -159,8 +156,6 @@ OSStatus ca_property_listener(AudioObjectPropertySelector selector,
for (int i = 0; i < n_addresses; i++) {
if (addresses[i].mSelector == selector) {
- ca_msg(MSGL_WARN, "event: property %s changed\n",
- fourcc_repr(talloc_ctx, selector));
if (data) *(volatile int *)data = 1;
break;
}
@@ -202,8 +197,8 @@ OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid) {
return noErr;
}
-static OSStatus ca_change_mixing(AudioDeviceID device, uint32_t val,
- bool *changed) {
+static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device,
+ uint32_t val, bool *changed) {
*changed = false;
AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
@@ -239,14 +234,14 @@ static OSStatus ca_change_mixing(AudioDeviceID device, uint32_t val,
return noErr;
}
-OSStatus ca_disable_mixing(AudioDeviceID device, bool *changed) {
- return ca_change_mixing(device, 0, changed);
+OSStatus ca_disable_mixing(struct ao *ao, AudioDeviceID device, bool *changed) {
+ return ca_change_mixing(ao, device, 0, changed);
}
-OSStatus ca_enable_mixing(AudioDeviceID device, bool changed) {
+OSStatus ca_enable_mixing(struct ao *ao, AudioDeviceID device, bool changed) {
if (changed) {
bool dont_care = false;
- return ca_change_mixing(device, 1, &dont_care);
+ return ca_change_mixing(ao, device, 1, &dont_care);
}
return noErr;
@@ -278,14 +273,14 @@ OSStatus ca_disable_device_listener(AudioDeviceID device, void *flag) {
return ca_change_device_listening(device, flag, false);
}
-bool ca_change_format(AudioStreamID stream,
+bool ca_change_format(struct ao *ao, AudioStreamID stream,
AudioStreamBasicDescription change_format)
{
OSStatus err = noErr;
AudioObjectPropertyAddress p_addr;
volatile int stream_format_changed = 0;
- ca_print_asbd("setting stream format:", &change_format);
+ ca_print_asbd(ao, "setting stream format:", &change_format);
/* Install the callback. */
p_addr = (AudioObjectPropertyAddress) {
@@ -317,13 +312,13 @@ bool ca_change_format(AudioStreamID stream,
if (stream_format_changed) {
stream_format_changed = 0;
} else {
- ca_msg(MSGL_V, "reached timeout\n");
+ MP_VERBOSE(ao, "reached timeout\n");
}
AudioStreamBasicDescription actual_format;
err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format);
- ca_print_asbd("actual format in use:", &actual_format);
+ ca_print_asbd(ao, "actual format in use:", &actual_format);
if (actual_format.mSampleRate == change_format.mSampleRate &&
actual_format.mFormatID == change_format.mFormatID &&
actual_format.mFramesPerPacket == change_format.mFramesPerPacket) {
@@ -341,33 +336,8 @@ bool ca_change_format(AudioStreamID stream,
return format_set;
}
-void ca_bitmaps_from_layouts(AudioChannelLayout *layouts, size_t n_layouts,
- uint32_t **bitmaps, size_t *n_bitmaps)
-{
- *n_bitmaps = 0;
- *bitmaps = talloc_array_size(NULL, sizeof(uint32_t), n_layouts);
-
- for (int i=0; i < n_layouts; i++) {
- uint32_t bitmap = 0;
-
- switch (layouts[i].mChannelLayoutTag) {
- case kAudioChannelLayoutTag_UseChannelBitmap:
- (*bitmaps)[(*n_bitmaps)++] = layouts[i].mChannelBitmap;
- break;
-
- case kAudioChannelLayoutTag_UseChannelDescriptions:
- if (ca_bitmap_from_ch_desc(&layouts[i], &bitmap))
- (*bitmaps)[(*n_bitmaps)++] = bitmap;
- break;
-
- default:
- if (ca_bitmap_from_ch_tag(&layouts[i], &bitmap))
- (*bitmaps)[(*n_bitmaps)++] = bitmap;
- }
- }
-}
-
-bool ca_bitmap_from_ch_desc(AudioChannelLayout *layout, uint32_t *bitmap)
+static bool ca_bitmap_from_ch_desc(struct ao *ao, AudioChannelLayout *layout,
+ uint32_t *bitmap)
{
// If the channel layout uses channel descriptions, from my
// exepriments there are there three possibile cases:
@@ -388,9 +358,8 @@ bool ca_bitmap_from_ch_desc(AudioChannelLayout *layout, uint32_t *bitmap)
if (label == kAudioChannelLabel_UseCoordinates ||
label == kAudioChannelLabel_Unknown ||
label > kAudioChannelLabel_TopBackRight) {
- ca_msg(MSGL_V,
- "channel label=%d unusable to build channel "
- "bitmap, skipping layout\n", label);
+ MP_VERBOSE(ao, "channel label=%d unusable to build channel "
+ "bitmap, skipping layout\n", label);
all_channels_valid = false;
} else {
*bitmap |= 1ULL << (label - 1);
@@ -400,7 +369,8 @@ bool ca_bitmap_from_ch_desc(AudioChannelLayout *layout, uint32_t *bitmap)
return all_channels_valid;
}
-bool ca_bitmap_from_ch_tag(AudioChannelLayout *layout, uint32_t *bitmap)
+static bool ca_bitmap_from_ch_tag(struct ao *ao, AudioChannelLayout *layout,
+ uint32_t *bitmap)
{
// This layout is defined exclusively by it's tag. Use the Audio
// Format Services API to try and convert it to a bitmap that
@@ -413,11 +383,37 @@ bool ca_bitmap_from_ch_tag(AudioChannelLayout *layout, uint32_t *bitmap)
sizeof(AudioChannelLayoutTag), &tag,
&bitmap_size, bitmap);
if (err != noErr) {
- ca_msg(MSGL_V,
- "channel layout tag=%d unusable to build channel "
- "bitmap, skipping layout\n", tag);
+ MP_VERBOSE(ao, "channel layout tag=%d unusable to build channel "
+ "bitmap, skipping layout\n", tag);
return false;
} else {
return true;
}
}
+
+void ca_bitmaps_from_layouts(struct ao *ao,
+ AudioChannelLayout *layouts, size_t n_layouts,
+ uint32_t **bitmaps, size_t *n_bitmaps)
+{
+ *n_bitmaps = 0;
+ *bitmaps = talloc_array_size(NULL, sizeof(uint32_t), n_layouts);
+
+ for (int i=0; i < n_layouts; i++) {
+ uint32_t bitmap = 0;
+
+ switch (layouts[i].mChannelLayoutTag) {
+ case kAudioChannelLayoutTag_UseChannelBitmap:
+ (*bitmaps)[(*n_bitmaps)++] = layouts[i].mChannelBitmap;
+ break;
+
+ case kAudioChannelLayoutTag_UseChannelDescriptions:
+ if (ca_bitmap_from_ch_desc(ao, &layouts[i], &bitmap))
+ (*bitmaps)[(*n_bitmaps)++] = bitmap;
+ break;
+
+ default:
+ if (ca_bitmap_from_ch_tag(ao, &layouts[i], &bitmap))
+ (*bitmaps)[(*n_bitmaps)++] = bitmap;
+ }
+ }
+}
diff --git a/audio/out/ao_coreaudio_utils.h b/audio/out/ao_coreaudio_utils.h
index 78e524fbd8..96383f9870 100644
--- a/audio/out/ao_coreaudio_utils.h
+++ b/audio/out/ao_coreaudio_utils.h
@@ -23,29 +23,34 @@
#include <inttypes.h>
#include <stdbool.h>
#include "core/mp_msg.h"
+#include "audio/out/ao.h"
-#define ca_msg(a, b ...) mp_msg(MSGT_AO, a, "AO: [coreaudio] " b)
#define CA_CFSTR_ENCODING kCFStringEncodingASCII
char *fourcc_repr(void *talloc_ctx, uint32_t code);
-bool check_ca_st(int level, OSStatus code, const char *message);
+bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message);
#define CHECK_CA_ERROR_L(label, message) \
do { \
- if (!check_ca_st(MSGL_ERR, err, message)) { \
+ if (!check_ca_st(ao, MSGL_ERR, err, message)) { \
goto label; \
} \
} while (0)
#define CHECK_CA_ERROR(message) CHECK_CA_ERROR_L(coreaudio_error, message)
-#define CHECK_CA_WARN(message) check_ca_st(MSGL_WARN, err, message)
+#define CHECK_CA_WARN(message) check_ca_st(ao, MSGL_WARN, err, message)
-void ca_print_asbd(const char *description,
+#define CHECK_CA_ERROR_SILENT_L(label) \
+ do { \
+ if (err != noErr) goto label; \
+ } while (0)
+
+void ca_print_asbd(struct ao *ao, const char *description,
const AudioStreamBasicDescription *asbd);
bool ca_format_is_digital(AudioStreamBasicDescription asbd);
-bool ca_stream_supports_digital(AudioStreamID stream);
-bool ca_device_supports_digital(AudioDeviceID device);
+bool ca_stream_supports_digital(struct ao *ao, AudioStreamID stream);
+bool ca_device_supports_digital(struct ao *ao, AudioDeviceID device);
OSStatus ca_property_listener(AudioObjectPropertySelector selector,
AudioObjectID object, uint32_t n_addresses,
@@ -62,17 +67,17 @@ OSStatus ca_device_listener(AudioObjectID object, uint32_t n_addresses,
OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid);
OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid);
-OSStatus ca_disable_mixing(AudioDeviceID device, bool *changed);
-OSStatus ca_enable_mixing(AudioDeviceID device, bool changed);
+OSStatus ca_disable_mixing(struct ao *ao, AudioDeviceID device, bool *changed);
+OSStatus ca_enable_mixing(struct ao *ao, AudioDeviceID device, bool changed);
+
OSStatus ca_enable_device_listener(AudioDeviceID device, void *flag);
OSStatus ca_disable_device_listener(AudioDeviceID device, void *flag);
-bool ca_change_format(AudioStreamID stream,
+bool ca_change_format(struct ao *ao, AudioStreamID stream,
AudioStreamBasicDescription change_format);
-bool ca_bitmap_from_ch_desc(AudioChannelLayout *layout, uint32_t *bitmap);
-bool ca_bitmap_from_ch_tag(AudioChannelLayout *layout, uint32_t *bitmap);
-void ca_bitmaps_from_layouts(AudioChannelLayout *layouts, size_t n_layouts,
+void ca_bitmaps_from_layouts(struct ao *ao,
+ AudioChannelLayout *layouts, size_t n_layouts,
uint32_t **bitmaps, size_t *n_bitmaps);
#endif /* MPV_COREAUDIO_UTILS_H */
diff --git a/audio/out/ao_wasapi.c b/audio/out/ao_wasapi.c
index 0e8d40f2bf..03a733ce33 100644
--- a/audio/out/ao_wasapi.c
+++ b/audio/out/ao_wasapi.c
@@ -294,8 +294,8 @@ static int set_ao_format(struct wasapi_state *state,
wformat.Format.wBitsPerSample, wformat.SubFormat.Data1 == 3);
if (wformat.SubFormat.Data1 != 1 && wformat.SubFormat.Data1 != 3) {
- mp_msg(MSGT_AO, MSGL_ERR, "ao-wasapi: unknown SubFormat %d\n",
- wformat.SubFormat.Data1);
+ mp_msg(MSGT_AO, MSGL_ERR, "ao-wasapi: unknown SubFormat %"PRIu32"\n",
+ (uint32_t)wformat.SubFormat.Data1);
return 0;
}
@@ -798,8 +798,8 @@ static int enumerate_devices(void) {
CoUninitialize();
return 0;
exit_label:
- mp_msg(MSGT_AO, MSGL_ERR, "Error enumerating devices: HRESULT %08x \"%s\"\n",
- hr, explain_err(hr));
+ mp_msg(MSGT_AO, MSGL_ERR, "Error enumerating devices: HRESULT %08"PRIx32" \"%s\"\n",
+ (uint32_t)hr, explain_err(hr));
CoUninitialize();
return 1;
}
@@ -1044,7 +1044,7 @@ static void thread_feed(wasapi_state *state,int force_feed)
return;
exit_label:
EnterCriticalSection(&state->print_lock);
- mp_msg(MSGT_AO, MSGL_ERR, "ao-wasapi: thread_feed fails with %"PRIx32"!\n", hr);
+ mp_msg(MSGT_AO, MSGL_ERR, "ao-wasapi: thread_feed fails with %"PRIx32"!\n", (uint32_t)hr);
LeaveCriticalSection(&state->print_lock);
return;
}
@@ -1107,7 +1107,7 @@ static void thread_uninit(wasapi_state *state)
ExitThread(0);
}
-static unsigned int __stdcall ThreadLoop(void *lpParameter)
+static DWORD __stdcall ThreadLoop(void *lpParameter)
{
struct ao *ao = lpParameter;
int feedwatch = 0;
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-18 20:57:17 +0000