mpv/audio/out/ao_coreaudio_utils.c

286 lines
9.4 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 General Public License as published by
* the Free Software Foundation; either version 2 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 General Public License for more details.
*
* You should have received a copy of the GNU 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 <CoreAudio/HostTime.h>
#include "audio/out/ao_coreaudio_utils.h"
#include "audio/out/ao_coreaudio_properties.h"
#include "osdep/timer.h"
#include "osdep/endian.h"
#include "audio/format.h"
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;
}
static bool ca_is_output_device(struct ao *ao, AudioDeviceID dev)
{
size_t n_buffers;
AudioBufferList *buffers;
const ca_scope scope = kAudioDevicePropertyStreamConfiguration;
CA_GET_ARY_O(dev, scope, &buffers, &n_buffers);
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);
talloc_steal(ta_ctx, name);
err = CA_GET_STR(devs[i], kAudioObjectPropertyName, &desc);
talloc_steal(ta_ctx, desc);
if (err != noErr)
desc = "Unknown";
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) {
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");
} 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;
err = CA_GET_STR(*device, kAudioObjectPropertyName, &desc);
CHECK_CA_WARN("could not get selected audio device name");
if (err == noErr) {
MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
desc, *device);
talloc_free(desc);
}
}
coreaudio_error:
return err;
}
char *fourcc_repr(void *talloc_ctx, uint32_t code)
{
// Extract FourCC letters from the uint32_t and finde out if it's a valid
// code that is made of letters.
unsigned char fcc[4] = {
(code >> 24) & 0xFF,
(code >> 16) & 0xFF,
(code >> 8) & 0xFF,
code & 0xFF,
};
bool valid_fourcc = true;
for (int i = 0; i < 4; i++) {
if (fcc[i] < 32 || fcc[i] >= 128)
valid_fourcc = false;
}
char *repr;
if (valid_fourcc)
repr = talloc_asprintf(talloc_ctx, "'%c%c%c%c'",
fcc[0], fcc[1], fcc[2], fcc[3]);
else
repr = talloc_asprintf(NULL, "%u", (unsigned int)code);
return repr;
}
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);
mp_msg(ao->log, level, "%s (%s)\n", message, error_string);
talloc_free(error_string);
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_FORMAT_IS_IEC61937(mp_format) ?
kAudioFormat60958AC3 :
kAudioFormatLinearPCM;
asbd->mChannelsPerFrame = num_channels;
asbd->mBitsPerChannel = af_fmt2bits(mp_format);
asbd->mFormatFlags = kAudioFormatFlagIsPacked;
if ((mp_format & AF_FORMAT_TYPE_MASK) == AF_FORMAT_F) {
asbd->mFormatFlags |= kAudioFormatFlagIsFloat;
} else if ((mp_format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI) {
asbd->mFormatFlags |= kAudioFormatFlagIsSignedInteger;
}
if (BYTE_ORDER == BIG_ENDIAN)
asbd->mFormatFlags |= kAudioFormatFlagIsBigEndian;
asbd->mFramesPerPacket = 1;
asbd->mBytesPerPacket = asbd->mBytesPerFrame =
asbd->mFramesPerPacket * asbd->mChannelsPerFrame *
(asbd->mBitsPerChannel / 8);
}
void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd)
{
ca_fill_asbd_raw(asbd, ao->format, ao->samplerate, ao->channels.num);
}
static bool ca_formatid_is_digital(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_digital(formatID) ? kAudioFormat60958AC3 : formatID;
}
bool ca_asbd_equals(const AudioStreamBasicDescription *a,
const AudioStreamBasicDescription *b)
{
int flags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsFloat |
kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsBigEndian;
return (a->mFormatFlags & flags) == (b->mFormatFlags & flags) &&
a->mBitsPerChannel == b->mBitsPerChannel &&
ca_normalize_formatid(a->mFormatID) ==
ca_normalize_formatid(b->mFormatID) &&
a->mBytesPerPacket == b->mBytesPerPacket;
}
// Return the AF_FORMAT_* (AF_FORMAT_S16 etc.) corresponding to the asbd.
int ca_asbd_to_mp_format(const AudioStreamBasicDescription *asbd)
{
for (int n = 0; af_fmtstr_table[n].format; n++) {
int mp_format = af_fmtstr_table[n].format;
AudioStreamBasicDescription mp_asbd = {0};
ca_fill_asbd_raw(&mp_asbd, mp_format, 0, asbd->mChannelsPerFrame);
if (ca_asbd_equals(&mp_asbd, asbd))
return mp_format;
}
return 0;
}
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);
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) : "unusable");
talloc_free(format);
}
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)
{
uint64_t out = AudioConvertHostTimeToNanos(ts->mHostTime);
uint64_t now = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
if (now > out)
return 0;
return (out - now) * 1e-3;
}