mpv/audio/out/ao_coreaudio_utils.c

555 lines
18 KiB
C

/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
/*
* This file contains functions interacting with the CoreAudio framework
* that are not specific to the AUHAL. These are split in a separate file for
* the sake of readability. In the future the could be used by other AOs based
* on CoreAudio but not the AUHAL (such as using AudioQueue services).
*/
#include "audio/out/ao_coreaudio_utils.h"
#include "osdep/timer.h"
#include "osdep/endian.h"
#include "osdep/semaphore.h"
#include "audio/format.h"
#if HAVE_COREAUDIO
#include "audio/out/ao_coreaudio_properties.h"
#include <CoreAudio/HostTime.h>
#else
#include <mach/mach_time.h>
#endif
CFStringRef cfstr_from_cstr(char *str)
{
return CFStringCreateWithCString(NULL, str, CA_CFSTR_ENCODING);
}
char *cfstr_get_cstr(CFStringRef cfstr)
{
CFIndex size =
CFStringGetMaximumSizeForEncoding(
CFStringGetLength(cfstr), CA_CFSTR_ENCODING) + 1;
char *buffer = talloc_zero_size(NULL, size);
CFStringGetCString(cfstr, buffer, size, CA_CFSTR_ENCODING);
return buffer;
}
#if HAVE_COREAUDIO
static bool ca_is_output_device(struct ao *ao, AudioDeviceID dev)
{
size_t n_buffers;
AudioBufferList *buffers;
const ca_scope scope = kAudioDevicePropertyStreamConfiguration;
OSStatus err = CA_GET_ARY_O(dev, scope, &buffers, &n_buffers);
if (err != noErr)
return false;
talloc_free(buffers);
return n_buffers > 0;
}
void ca_get_device_list(struct ao *ao, struct ao_device_list *list)
{
AudioDeviceID *devs;
size_t n_devs;
OSStatus err =
CA_GET_ARY(kAudioObjectSystemObject, kAudioHardwarePropertyDevices,
&devs, &n_devs);
CHECK_CA_ERROR("Failed to get list of output devices.");
for (int i = 0; i < n_devs; i++) {
if (!ca_is_output_device(ao, devs[i]))
continue;
void *ta_ctx = talloc_new(NULL);
char *name;
char *desc;
err = CA_GET_STR(devs[i], kAudioDevicePropertyDeviceUID, &name);
if (err != noErr) {
MP_VERBOSE(ao, "skipping device %d, which has no UID\n", i);
talloc_free(ta_ctx);
continue;
}
talloc_steal(ta_ctx, name);
err = CA_GET_STR(devs[i], kAudioObjectPropertyName, &desc);
if (err != noErr)
desc = talloc_strdup(NULL, "Unknown");
talloc_steal(ta_ctx, desc);
ao_device_list_add(list, ao, &(struct ao_device_desc){name, desc});
talloc_free(ta_ctx);
}
talloc_free(devs);
coreaudio_error:
return;
}
OSStatus ca_select_device(struct ao *ao, char* name, AudioDeviceID *device)
{
OSStatus err = noErr;
*device = kAudioObjectUnknown;
if (name && name[0]) {
CFStringRef uid = cfstr_from_cstr(name);
AudioValueTranslation v = (AudioValueTranslation) {
.mInputData = &uid,
.mInputDataSize = sizeof(CFStringRef),
.mOutputData = device,
.mOutputDataSize = sizeof(*device),
};
uint32_t size = sizeof(AudioValueTranslation);
AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
.mSelector = kAudioHardwarePropertyDeviceForUID,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster,
};
err = AudioObjectGetPropertyData(
kAudioObjectSystemObject, &p_addr, 0, 0, &size, &v);
CFRelease(uid);
CHECK_CA_ERROR("unable to query for device UID");
uint32_t is_alive = 1;
err = CA_GET(*device, kAudioDevicePropertyDeviceIsAlive, &is_alive);
CHECK_CA_ERROR("could not check whether device is alive (invalid device?)");
if (!is_alive)
MP_WARN(ao, "device is not alive!\n");
} else {
// device not set by user, get the default one
err = CA_GET(kAudioObjectSystemObject,
kAudioHardwarePropertyDefaultOutputDevice,
device);
CHECK_CA_ERROR("could not get default audio device");
}
if (mp_msg_test(ao->log, MSGL_V)) {
char *desc;
OSStatus err2 = CA_GET_STR(*device, kAudioObjectPropertyName, &desc);
if (err2 == noErr) {
MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
desc, *device);
talloc_free(desc);
}
}
coreaudio_error:
return err;
}
#endif
bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message)
{
if (code == noErr) return true;
mp_msg(ao->log, level, "%s (%s/%d)\n", message, mp_tag_str(code), (int)code);
return false;
}
static void ca_fill_asbd_raw(AudioStreamBasicDescription *asbd, int mp_format,
int samplerate, int num_channels)
{
asbd->mSampleRate = samplerate;
// Set "AC3" for other spdif formats too - unknown if that works.
asbd->mFormatID = af_fmt_is_spdif(mp_format) ?
kAudioFormat60958AC3 :
kAudioFormatLinearPCM;
asbd->mChannelsPerFrame = num_channels;
asbd->mBitsPerChannel = af_fmt_to_bytes(mp_format) * 8;
asbd->mFormatFlags = kAudioFormatFlagIsPacked;
int channels_per_buffer = num_channels;
if (af_fmt_is_planar(mp_format)) {
asbd->mFormatFlags |= kAudioFormatFlagIsNonInterleaved;
channels_per_buffer = 1;
}
if (af_fmt_is_float(mp_format)) {
asbd->mFormatFlags |= kAudioFormatFlagIsFloat;
} else if (!af_fmt_is_unsigned(mp_format)) {
asbd->mFormatFlags |= kAudioFormatFlagIsSignedInteger;
}
if (BYTE_ORDER == BIG_ENDIAN)
asbd->mFormatFlags |= kAudioFormatFlagIsBigEndian;
asbd->mFramesPerPacket = 1;
asbd->mBytesPerPacket = asbd->mBytesPerFrame =
asbd->mFramesPerPacket * channels_per_buffer *
(asbd->mBitsPerChannel / 8);
}
void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd)
{
ca_fill_asbd_raw(asbd, ao->format, ao->samplerate, ao->channels.num);
}
bool ca_formatid_is_compressed(uint32_t formatid)
{
switch (formatid)
case 'IAC3':
case 'iac3':
case kAudioFormat60958AC3:
case kAudioFormatAC3:
return true;
return false;
}
// This might be wrong, but for now it's sufficient for us.
static uint32_t ca_normalize_formatid(uint32_t formatID)
{
return ca_formatid_is_compressed(formatID) ? kAudioFormat60958AC3 : formatID;
}
bool ca_asbd_equals(const AudioStreamBasicDescription *a,
const AudioStreamBasicDescription *b)
{
int flags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsFloat |
kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsBigEndian;
bool spdif = ca_formatid_is_compressed(a->mFormatID) &&
ca_formatid_is_compressed(b->mFormatID);
return (a->mFormatFlags & flags) == (b->mFormatFlags & flags) &&
a->mBitsPerChannel == b->mBitsPerChannel &&
ca_normalize_formatid(a->mFormatID) ==
ca_normalize_formatid(b->mFormatID) &&
(spdif || a->mBytesPerPacket == b->mBytesPerPacket) &&
(spdif || a->mChannelsPerFrame == b->mChannelsPerFrame) &&
a->mSampleRate == b->mSampleRate;
}
// Return the AF_FORMAT_* (AF_FORMAT_S16 etc.) corresponding to the asbd.
int ca_asbd_to_mp_format(const AudioStreamBasicDescription *asbd)
{
for (int fmt = 1; fmt < AF_FORMAT_COUNT; fmt++) {
AudioStreamBasicDescription mp_asbd = {0};
ca_fill_asbd_raw(&mp_asbd, fmt, asbd->mSampleRate, asbd->mChannelsPerFrame);
if (ca_asbd_equals(&mp_asbd, asbd))
return af_fmt_is_spdif(fmt) ? AF_FORMAT_S_AC3 : fmt;
}
return 0;
}
void ca_print_asbd(struct ao *ao, const char *description,
const AudioStreamBasicDescription *asbd)
{
uint32_t flags = asbd->mFormatFlags;
char *format = mp_tag_str(asbd->mFormatID);
int mpfmt = ca_asbd_to_mp_format(asbd);
MP_VERBOSE(ao,
"%s %7.1fHz %" PRIu32 "bit %s "
"[%" PRIu32 "][%" PRIu32 "bpp][%" PRIu32 "fbp]"
"[%" PRIu32 "bpf][%" PRIu32 "ch] "
"%s %s %s%s%s%s (%s)\n",
description, asbd->mSampleRate, asbd->mBitsPerChannel, format,
asbd->mFormatFlags, asbd->mBytesPerPacket, asbd->mFramesPerPacket,
asbd->mBytesPerFrame, asbd->mChannelsPerFrame,
(flags & kAudioFormatFlagIsFloat) ? "float" : "int",
(flags & kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
(flags & kAudioFormatFlagIsSignedInteger) ? "S" : "U",
(flags & kAudioFormatFlagIsPacked) ? " packed" : "",
(flags & kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
(flags & kAudioFormatFlagIsNonInterleaved) ? " P" : "",
mpfmt ? af_fmt_to_str(mpfmt) : "-");
}
// Return whether new is an improvement over old. Assume a higher value means
// better quality, and we always prefer the value closest to the requested one,
// which is still larger than the requested one.
// Equal values prefer the new one (so ca_asbd_is_better() checks other params).
static bool value_is_better(double req, double old, double new)
{
if (new >= req) {
return old < req || new <= old;
} else {
return old < req && new >= old;
}
}
// Return whether new is an improvement over old (req is the requested format).
bool ca_asbd_is_better(AudioStreamBasicDescription *req,
AudioStreamBasicDescription *old,
AudioStreamBasicDescription *new)
{
if (new->mChannelsPerFrame > MP_NUM_CHANNELS)
return false;
if (old->mChannelsPerFrame > MP_NUM_CHANNELS)
return true;
if (req->mFormatID != new->mFormatID)
return false;
if (req->mFormatID != old->mFormatID)
return true;
if (!value_is_better(req->mBitsPerChannel, old->mBitsPerChannel,
new->mBitsPerChannel))
return false;
if (!value_is_better(req->mSampleRate, old->mSampleRate, new->mSampleRate))
return false;
if (!value_is_better(req->mChannelsPerFrame, old->mChannelsPerFrame,
new->mChannelsPerFrame))
return false;
return true;
}
int64_t ca_frames_to_us(struct ao *ao, uint32_t frames)
{
return frames / (float) ao->samplerate * 1e6;
}
int64_t ca_get_latency(const AudioTimeStamp *ts)
{
#if HAVE_COREAUDIO
uint64_t out = AudioConvertHostTimeToNanos(ts->mHostTime);
uint64_t now = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
if (now > out)
return 0;
return (out - now) * 1e-3;
#else
static mach_timebase_info_data_t timebase;
if (timebase.denom == 0)
mach_timebase_info(&timebase);
uint64_t out = ts->mHostTime;
uint64_t now = mach_absolute_time();
if (now > out)
return 0;
return (out - now) * timebase.numer / timebase.denom / 1e3;
#endif
}
#if HAVE_COREAUDIO
bool ca_stream_supports_compressed(struct ao *ao, AudioStreamID stream)
{
AudioStreamRangedDescription *formats = NULL;
size_t n_formats;
OSStatus err =
CA_GET_ARY(stream, kAudioStreamPropertyAvailablePhysicalFormats,
&formats, &n_formats);
CHECK_CA_ERROR("Could not get number of stream formats.");
for (int i = 0; i < n_formats; i++) {
AudioStreamBasicDescription asbd = formats[i].mFormat;
ca_print_asbd(ao, "- ", &asbd);
if (ca_formatid_is_compressed(asbd.mFormatID)) {
talloc_free(formats);
return true;
}
}
talloc_free(formats);
coreaudio_error:
return false;
}
OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid)
{
*pid = getpid();
OSStatus err = CA_SET(device, kAudioDevicePropertyHogMode, pid);
if (err != noErr)
*pid = -1;
return err;
}
OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid)
{
if (*pid == getpid()) {
*pid = -1;
return CA_SET(device, kAudioDevicePropertyHogMode, &pid);
}
return noErr;
}
static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device,
uint32_t val, bool *changed)
{
*changed = false;
AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
.mSelector = kAudioDevicePropertySupportsMixing,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster,
};
if (AudioObjectHasProperty(device, &p_addr)) {
OSStatus err;
Boolean writeable = 0;
err = CA_SETTABLE(device, kAudioDevicePropertySupportsMixing,
&writeable);
if (!CHECK_CA_WARN("can't tell if mixing property is settable")) {
return err;
}
if (!writeable)
return noErr;
err = CA_SET(device, kAudioDevicePropertySupportsMixing, &val);
if (err != noErr)
return err;
if (!CHECK_CA_WARN("can't set mix mode")) {
return err;
}
*changed = true;
}
return noErr;
}
OSStatus ca_disable_mixing(struct ao *ao, AudioDeviceID device, bool *changed)
{
return ca_change_mixing(ao, device, 0, changed);
}
OSStatus ca_enable_mixing(struct ao *ao, AudioDeviceID device, bool changed)
{
if (changed) {
bool dont_care = false;
return ca_change_mixing(ao, device, 1, &dont_care);
}
return noErr;
}
int64_t ca_get_device_latency_us(struct ao *ao, AudioDeviceID device)
{
uint32_t latency_frames = 0;
uint32_t latency_properties[] = {
kAudioDevicePropertyLatency,
kAudioDevicePropertyBufferFrameSize,
kAudioDevicePropertySafetyOffset,
};
for (int n = 0; n < MP_ARRAY_SIZE(latency_properties); n++) {
uint32_t temp;
OSStatus err = CA_GET_O(device, latency_properties[n], &temp);
CHECK_CA_WARN("cannot get device latency");
if (err == noErr) {
latency_frames += temp;
MP_VERBOSE(ao, "Latency property %s: %d frames\n",
mp_tag_str(latency_properties[n]), (int)temp);
}
}
double sample_rate = ao->samplerate;
OSStatus err = CA_GET_O(device, kAudioDevicePropertyNominalSampleRate,
&sample_rate);
CHECK_CA_WARN("cannot get device sample rate, falling back to AO sample rate!");
if (err == noErr) {
MP_VERBOSE(ao, "Device sample rate: %f\n", sample_rate);
}
return latency_frames / sample_rate * 1e6;
}
static OSStatus ca_change_format_listener(
AudioObjectID object, uint32_t n_addresses,
const AudioObjectPropertyAddress addresses[],
void *data)
{
sem_t *sem = data;
sem_post(sem);
return noErr;
}
bool ca_change_physical_format_sync(struct ao *ao, AudioStreamID stream,
AudioStreamBasicDescription change_format)
{
OSStatus err = noErr;
bool format_set = false;
ca_print_asbd(ao, "setting stream physical format:", &change_format);
sem_t wakeup;
if (sem_init(&wakeup, 0, 0)) {
MP_WARN(ao, "OOM\n");
return false;
}
AudioStreamBasicDescription prev_format;
err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &prev_format);
CHECK_CA_ERROR("can't get current physical format");
ca_print_asbd(ao, "format in use before switching:", &prev_format);
/* Install the callback. */
AudioObjectPropertyAddress p_addr = {
.mSelector = kAudioStreamPropertyPhysicalFormat,
.mScope = kAudioObjectPropertyScopeGlobal,
.mElement = kAudioObjectPropertyElementMaster,
};
err = AudioObjectAddPropertyListener(stream, &p_addr,
ca_change_format_listener,
&wakeup);
CHECK_CA_ERROR("can't add property listener during format change");
/* Change the format. */
err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &change_format);
CHECK_CA_WARN("error changing physical format");
/* The AudioStreamSetProperty is not only asynchronous,
* it is also not Atomic, in its behaviour. */
struct timespec timeout = mp_rel_time_to_timespec(2.0);
AudioStreamBasicDescription actual_format = {0};
while (1) {
err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format);
if (!CHECK_CA_WARN("could not retrieve physical format"))
break;
format_set = ca_asbd_equals(&change_format, &actual_format);
if (format_set)
break;
if (sem_timedwait(&wakeup, &timeout)) {
MP_VERBOSE(ao, "reached timeout\n");
break;
}
}
ca_print_asbd(ao, "actual format in use:", &actual_format);
if (!format_set) {
MP_WARN(ao, "changing physical format failed\n");
// Some drivers just fuck up and get into a broken state. Restore the
// old format in this case.
err = CA_SET(stream, kAudioStreamPropertyPhysicalFormat, &prev_format);
CHECK_CA_WARN("error restoring physical format");
}
err = AudioObjectRemovePropertyListener(stream, &p_addr,
ca_change_format_listener,
&wakeup);
CHECK_CA_ERROR("can't remove property listener");
coreaudio_error:
sem_destroy(&wakeup);
return format_set;
}
#endif