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gstreamer/subprojects/gst-plugins-good/sys/osxaudio/gstosxcoreaudio.c
Piotr Brzeziński a88eda66d2 osxaudio: Always expose max amount of channels with no positions 
For outputs with a high number of channels, macOS has a bug where
initially CoreAudio will report incorrect positions for all channels,
but after you run Audio MIDI Setup and configure the speaker layout
there, macOS will always report those few as positioned, with no option
to revert that (other than deleting some internal files).

In such scenario our code would just ignore all the unpositioned
channels. Since you can only position max. 16 channels in macOS, if you
had more on your output device, those would be unusable.

This commit makes sure that in addition to the usual positioned layout
(if there is one), we will expose caps for a no-positions layout that
always has the maximum amount of channels available.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8311>
2025-01-30 09:53:15 +00:00

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/*
* GStreamer
* Copyright (C) 2012-2013 Fluendo S.A. <support@fluendo.com>
* Authors: Josep Torra Vallès <josep@fluendo.com>
* Andoni Morales Alastruey <amorales@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*
*/
#include "gstosxcoreaudio.h"
#include "gstosxcoreaudiocommon.h"
GST_DEBUG_CATEGORY (osx_coreaudio_debug);
#define GST_CAT_DEFAULT osx_coreaudio_debug
G_DEFINE_TYPE (GstCoreAudio, gst_core_audio, G_TYPE_OBJECT);
#ifdef HAVE_IOS
#include "gstosxcoreaudioremoteio.c"
#else
#include "gstosxcoreaudiohal.c"
#include <CoreAudio/CoreAudio.h>
#endif
enum
{
PROP_0,
PROP_DEVICE,
PROP_IS_SRC,
PROP_CONFIGURE_SESSION,
};
static void gst_core_audio_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_core_audio_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void
gst_core_audio_finalize (GObject * object)
{
GstCoreAudio *core_audio = GST_CORE_AUDIO (object);
g_mutex_clear (&core_audio->timing_lock);
g_free (core_audio->unique_id);
G_OBJECT_CLASS (gst_core_audio_parent_class)->finalize (object);
}
static void
gst_core_audio_class_init (GstCoreAudioClass * klass)
{
GObjectClass *object_klass = G_OBJECT_CLASS (klass);
object_klass->finalize = gst_core_audio_finalize;
object_klass->set_property = gst_core_audio_set_property;
object_klass->get_property = gst_core_audio_get_property;
g_object_class_install_property (object_klass, PROP_DEVICE,
g_param_spec_int ("device", "Device ID", "Device ID of input device",
0, G_MAXINT, 0,
G_PARAM_STATIC_STRINGS | G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY));
g_object_class_install_property (object_klass, PROP_IS_SRC,
g_param_spec_boolean ("is-src", "Is source", "Is a source device",
FALSE,
G_PARAM_STATIC_STRINGS | G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY));
#ifdef HAVE_IOS
g_object_class_install_property (object_klass, PROP_CONFIGURE_SESSION,
g_param_spec_boolean ("configure-session",
"Enable automatic AVAudioSession setup",
"Auto-configure the AVAudioSession for audio playback/capture", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT_ONLY));
#endif
}
static void
gst_core_audio_init (GstCoreAudio * core_audio)
{
core_audio->is_passthrough = FALSE;
core_audio->device_id = kAudioDeviceUnknown;
core_audio->unique_id = NULL;
core_audio->is_src = FALSE;
core_audio->audiounit = NULL;
core_audio->cached_caps = NULL;
core_audio->cached_caps_valid = FALSE;
#ifndef HAVE_IOS
core_audio->hog_pid = -1;
core_audio->disabled_mixing = FALSE;
#else
core_audio->configure_session = FALSE;
#endif
mach_timebase_info (&core_audio->timebase);
g_mutex_init (&core_audio->timing_lock);
}
static void
gst_core_audio_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstCoreAudio *self = GST_CORE_AUDIO (object);
switch (prop_id) {
case PROP_IS_SRC:
self->is_src = g_value_get_boolean (value);
break;
case PROP_DEVICE:
self->device_id = g_value_get_int (value);
break;
#ifdef HAVE_IOS
case PROP_CONFIGURE_SESSION:
self->configure_session = g_value_get_boolean (value);
break;
#endif
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_core_audio_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstCoreAudio *self = GST_CORE_AUDIO (object);
switch (prop_id) {
case PROP_IS_SRC:
g_value_set_boolean (value, self->is_src);
break;
case PROP_DEVICE:
g_value_set_int (value, self->device_id);
break;
#ifdef HAVE_IOS
case PROP_CONFIGURE_SESSION:
g_value_set_boolean (value, self->configure_session);
break;
#endif
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
_is_outer_scope (AudioUnitScope scope, AudioUnitElement element)
{
return
(scope == kAudioUnitScope_Input && element == 1) ||
(scope == kAudioUnitScope_Output && element == 0);
}
static void
_audio_unit_property_listener (void *inRefCon, AudioUnit inUnit,
AudioUnitPropertyID inID, AudioUnitScope inScope,
AudioUnitElement inElement)
{
GstCoreAudio *core_audio;
core_audio = GST_CORE_AUDIO (inRefCon);
g_assert (inUnit == core_audio->audiounit);
switch (inID) {
case kAudioUnitProperty_AudioChannelLayout:
case kAudioUnitProperty_StreamFormat:
if (_is_outer_scope (inScope, inElement)) {
/* We don't push gst_event_new_caps here (for src),
* nor gst_event_new_reconfigure (for sink), since Core Audio continues
* to happily function with the old format, doing conversion/resampling
* as needed.
* This merely "refreshes" our PREFERRED caps. */
/* This function is called either from a Core Audio thread
* or as a result of a Core Audio API (e.g. AudioUnitInitialize)
* from our own thread. In the latter case, osxbuf can be
* already locked (GStreamer's mutex is not recursive).
* For this reason we use a boolean flag instead of nullifying
* cached_caps. */
core_audio->cached_caps_valid = FALSE;
}
break;
}
}
static GstClockTime
_current_time_ns (GstCoreAudio * core_audio)
{
guint64 mach_t = mach_absolute_time ();
return gst_util_uint64_scale (mach_t, core_audio->timebase.numer,
core_audio->timebase.denom);
}
static GstClockTime
_host_time_to_ns (GstCoreAudio * core_audio, uint64_t host_time)
{
return gst_util_uint64_scale (host_time, core_audio->timebase.numer,
core_audio->timebase.denom);
}
/**************************
* Public API *
*************************/
gboolean
gst_core_audio_close (GstCoreAudio * core_audio)
{
OSStatus status;
/* Uninitialize the AudioUnit */
status = AudioUnitUninitialize (core_audio->audiounit);
if (status) {
GST_ERROR_OBJECT (core_audio, "Failed to uninitialize AudioUnit: %d",
(int) status);
return FALSE;
}
AudioUnitRemovePropertyListenerWithUserData (core_audio->audiounit,
kAudioUnitProperty_AudioChannelLayout, _audio_unit_property_listener,
core_audio);
AudioUnitRemovePropertyListenerWithUserData (core_audio->audiounit,
kAudioUnitProperty_StreamFormat, _audio_unit_property_listener,
core_audio);
/* core_audio->osxbuf is already locked at this point */
core_audio->cached_caps_valid = FALSE;
gst_caps_replace (&core_audio->cached_caps, NULL);
AudioComponentInstanceDispose (core_audio->audiounit);
core_audio->audiounit = NULL;
return TRUE;
}
gboolean
gst_core_audio_open (GstCoreAudio * core_audio)
{
OSStatus status;
/* core_audio->osxbuf is already locked at this point */
core_audio->cached_caps_valid = FALSE;
gst_caps_replace (&core_audio->cached_caps, NULL);
if (!gst_core_audio_open_impl (core_audio))
return FALSE;
/* Add property listener */
status = AudioUnitAddPropertyListener (core_audio->audiounit,
kAudioUnitProperty_AudioChannelLayout, _audio_unit_property_listener,
core_audio);
if (status != noErr) {
GST_ERROR_OBJECT (core_audio, "Failed to add audio channel layout property "
"listener for AudioUnit: %d", (int) status);
}
status = AudioUnitAddPropertyListener (core_audio->audiounit,
kAudioUnitProperty_StreamFormat, _audio_unit_property_listener,
core_audio);
if (status != noErr) {
GST_ERROR_OBJECT (core_audio, "Failed to add stream format property "
"listener for AudioUnit: %d", (int) status);
}
/* Initialize the AudioUnit. We keep the audio unit initialized early so that
* we can probe the underlying device. */
status = AudioUnitInitialize (core_audio->audiounit);
if (status) {
GST_ERROR_OBJECT (core_audio, "Failed to initialize AudioUnit: %d",
(int) status);
return FALSE;
}
return TRUE;
}
gboolean
gst_core_audio_start_processing (GstCoreAudio * core_audio)
{
return gst_core_audio_start_processing_impl (core_audio);
}
gboolean
gst_core_audio_pause_processing (GstCoreAudio * core_audio)
{
return gst_core_audio_pause_processing_impl (core_audio);
}
gboolean
gst_core_audio_stop_processing (GstCoreAudio * core_audio)
{
return gst_core_audio_stop_processing_impl (core_audio);
}
gboolean
gst_core_audio_get_samples_and_latency (GstCoreAudio * core_audio,
gdouble rate, guint * samples, gdouble * latency)
{
uint64_t now_ns = _current_time_ns (core_audio);
gboolean ret = gst_core_audio_get_samples_and_latency_impl (core_audio, rate,
samples, latency);
if (!ret)
return FALSE;
CORE_AUDIO_TIMING_LOCK (core_audio);
uint32_t samples_remain = 0;
uint64_t anchor_ns = core_audio->anchor_hosttime_ns;
if (core_audio->is_src) {
int64_t captured_ns =
core_audio->rate_scalar * (int64_t) (now_ns - anchor_ns);
/* src, the anchor time is the timestamp of the first sample in the last
* packet received, and we increment up from there, unless the device gets stopped. */
if (captured_ns > 0) {
if (core_audio->io_proc_active) {
samples_remain = (uint32_t) (captured_ns * rate / GST_SECOND);
} else {
samples_remain = core_audio->anchor_pend_samples;
}
} else {
/* Time went backward. This shouldn't happen for sources, but report something anyway */
samples_remain =
(uint32_t) (-captured_ns * rate / GST_SECOND) +
core_audio->anchor_pend_samples;
}
GST_DEBUG_OBJECT (core_audio,
"now_ns %" G_GUINT64_FORMAT " anchor %" G_GUINT64_FORMAT " elapsed ns %"
G_GINT64_FORMAT " rate %f captured_ns %" G_GINT64_FORMAT
" anchor_pend_samples %u samples_remain %u", now_ns, anchor_ns,
now_ns - anchor_ns, rate, captured_ns, core_audio->anchor_pend_samples,
samples_remain);
} else {
/* Sink, the anchor time is the time the most recent buffer will commence play out,
* and we count down to 0 for unplayed samples beyond that */
int64_t unplayed_ns =
core_audio->rate_scalar * (int64_t) (anchor_ns - now_ns);
if (unplayed_ns > 0) {
samples_remain =
(uint32_t) (unplayed_ns * rate / GST_SECOND) +
core_audio->anchor_pend_samples;
} else {
uint32_t samples_played = (uint32_t) (-unplayed_ns * rate / GST_SECOND);
if (samples_played < core_audio->anchor_pend_samples) {
samples_remain = core_audio->anchor_pend_samples - samples_played;
}
}
GST_DEBUG_OBJECT (core_audio,
"now_ns %" G_GUINT64_FORMAT " anchor %" G_GUINT64_FORMAT " elapsed ns %"
G_GINT64_FORMAT " rate %f unplayed_ns %" G_GINT64_FORMAT
" anchor_pend_samples %u", now_ns, anchor_ns, now_ns - anchor_ns, rate,
unplayed_ns, core_audio->anchor_pend_samples);
}
CORE_AUDIO_TIMING_UNLOCK (core_audio);
GST_DEBUG_OBJECT (core_audio, "samples = %u latency %f", samples_remain,
*latency);
*samples = samples_remain;
return TRUE;
}
void
gst_core_audio_update_timing (GstCoreAudio * core_audio,
const AudioTimeStamp * inTimeStamp, unsigned int inNumberFrames)
{
AudioTimeStampFlags target_flags =
kAudioTimeStampSampleHostTimeValid | kAudioTimeStampRateScalarValid;
if ((inTimeStamp->mFlags & target_flags) == target_flags) {
core_audio->anchor_hosttime_ns =
_host_time_to_ns (core_audio, inTimeStamp->mHostTime);
core_audio->anchor_pend_samples = inNumberFrames;
core_audio->rate_scalar = inTimeStamp->mRateScalar;
GST_DEBUG_OBJECT (core_audio,
"anchor hosttime_ns %" G_GUINT64_FORMAT
" scalar_rate %f anchor_pend_samples %u",
core_audio->anchor_hosttime_ns,
core_audio->rate_scalar, core_audio->anchor_pend_samples);
}
}
gboolean
gst_core_audio_initialize (GstCoreAudio * core_audio,
AudioStreamBasicDescription format, GstCaps * caps,
guint32 frames_per_packet, gboolean is_passthrough)
{
GST_DEBUG_OBJECT (core_audio,
"Initializing: passthrough:%d caps:%" GST_PTR_FORMAT, is_passthrough,
caps);
if (!gst_core_audio_initialize_impl (core_audio, format, caps,
is_passthrough, &frames_per_packet)) {
return FALSE;
}
if (core_audio->is_src) {
/* create AudioBufferList needed for recording */
core_audio->recBufferSize = frames_per_packet * format.mBytesPerFrame;
GST_DEBUG_OBJECT (core_audio,
"Allocating record buffers %u bytes %u frames",
core_audio->recBufferSize, frames_per_packet);
core_audio->recBufferList =
buffer_list_alloc (format.mChannelsPerFrame, core_audio->recBufferSize,
/* Currently always TRUE (i.e. interleaved) */
!(format.mFormatFlags & kAudioFormatFlagIsNonInterleaved));
}
return TRUE;
}
void
gst_core_audio_uninitialize (GstCoreAudio * core_audio)
{
buffer_list_free (core_audio->recBufferList);
core_audio->recBufferList = NULL;
}
void
gst_core_audio_set_volume (GstCoreAudio * core_audio, gfloat volume)
{
AudioUnitSetParameter (core_audio->audiounit, kHALOutputParam_Volume,
kAudioUnitScope_Global, 0, (float) volume, 0);
}
gboolean
gst_core_audio_select_device (GstCoreAudio * core_audio)
{
gboolean ret = gst_core_audio_select_device_impl (core_audio);
#ifndef HAVE_IOS
if (core_audio->device_id != kAudioDeviceUnknown)
core_audio->unique_id =
gst_core_audio_device_get_prop (core_audio->device_id,
kAudioDevicePropertyDeviceUID);
#endif
return ret;
}
void
gst_core_audio_init_debug (void)
{
GST_DEBUG_CATEGORY_INIT (osx_coreaudio_debug, "osxaudio", 0,
"OSX Audio Elements");
}
gboolean
gst_core_audio_audio_device_is_spdif_avail (AudioDeviceID device_id)
{
return gst_core_audio_audio_device_is_spdif_avail_impl (device_id);
}
/* Does the channel have at least one positioned channel?
* (GStreamer is more strict than Core Audio, in that it requires either
* all channels to be positioned, or all unpositioned.) */
static gboolean
_is_core_audio_layout_positioned (AudioChannelLayout * layout)
{
guint i;
g_assert (layout->mChannelLayoutTag ==
kAudioChannelLayoutTag_UseChannelDescriptions);
for (i = 0; i < layout->mNumberChannelDescriptions; ++i) {
GstAudioChannelPosition p =
gst_core_audio_channel_label_to_gst
(layout->mChannelDescriptions[i].mChannelLabel, i, FALSE);
if (p >= 0) /* not special positition */
return TRUE;
}
return FALSE;
}
static gboolean
_core_audio_has_invalid_channel_labels (AudioChannelLayout * layout)
{
guint i;
g_assert (layout->mChannelLayoutTag ==
kAudioChannelLayoutTag_UseChannelDescriptions);
for (i = 0; i < layout->mNumberChannelDescriptions; ++i) {
/* Let's use our mapping to judge whether the value is valid.
* It doesn't support all of the defined positions, but the missing ones
* aren't useful to us anyway. */
GstAudioChannelPosition p =
gst_core_audio_channel_label_to_gst
(layout->mChannelDescriptions[i].mChannelLabel, i, FALSE);
if (p == GST_AUDIO_CHANNEL_POSITION_INVALID)
return TRUE;
}
return FALSE;
}
static void
_core_audio_parse_channel_descriptions (AudioChannelLayout * layout,
guint * channels, guint64 * channel_mask, GstAudioChannelPosition * pos)
{
gboolean positioned;
guint i;
g_assert (layout->mChannelLayoutTag ==
kAudioChannelLayoutTag_UseChannelDescriptions);
/* For >16ch devices, CoreAudio can give out completely incorrect
* channel positions by default - instead of using kAudioChannelLabel_Discrete_X,
* it just returns incrementing values starting from 0, some of which are not
* valid if you check against the CoreAudioBaseTypes.h header.
* If such case is detected, let's just swap all positions to Discrete,
* which map to GST_AUDIO_CHANNEL_POSITION_NONE. */
if (_core_audio_has_invalid_channel_labels (layout)) {
GST_DEBUG
("Invalid channel positions given by CoreAudio, setting all to unpositioned");
if (pos) {
for (i = 0; i < layout->mNumberChannelDescriptions; ++i)
pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
}
*channels = layout->mNumberChannelDescriptions;
*channel_mask = 0;
return;
}
positioned = _is_core_audio_layout_positioned (layout);
*channel_mask = 0;
/* Go over all labels, either taking only positioned or only
* unpositioned channels, up to GST_OSX_AUDIO_MAX_CHANNEL channels.
*
* The resulting 'pos' array will contain either:
* - only regular (>= 0) positions
* - only GST_AUDIO_CHANNEL_POSITION_NONE positions
* in a compact form, skipping over all unsupported positions.
*/
*channels = 0;
for (i = 0; i < layout->mNumberChannelDescriptions; ++i) {
GstAudioChannelPosition p =
gst_core_audio_channel_label_to_gst
(layout->mChannelDescriptions[i].mChannelLabel, i, TRUE);
/* In positioned layouts, skip all unpositioned channels.
* In unpositioned layouts, skip all invalid channels. */
if ((positioned && p >= 0) ||
(!positioned && p == GST_AUDIO_CHANNEL_POSITION_NONE)) {
if (pos)
pos[*channels] = p;
*channel_mask |= G_GUINT64_CONSTANT (1) << p;
++(*channels);
if (*channels == GST_OSX_AUDIO_MAX_CHANNEL)
break; /* not to overflow */
}
}
}
gboolean
gst_core_audio_parse_channel_layout (AudioChannelLayout * layout,
guint * channels, guint64 * channel_mask, GstAudioChannelPosition * pos)
{
g_assert (channels != NULL);
g_assert (channel_mask != NULL);
g_assert (layout != NULL);
if (layout->mChannelLayoutTag ==
kAudioChannelLayoutTag_UseChannelDescriptions) {
switch (layout->mNumberChannelDescriptions) {
case 0:
if (pos)
pos[0] = GST_AUDIO_CHANNEL_POSITION_NONE;
*channels = 0;
*channel_mask = 0;
return TRUE;
case 1:
if (pos)
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
*channels = 1;
*channel_mask = 0;
return TRUE;
case 2:
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
*channels = 2;
*channel_mask =
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
return TRUE;
default:
_core_audio_parse_channel_descriptions (layout, channels, channel_mask,
pos);
return TRUE;
}
} else if (layout->mChannelLayoutTag == kAudioChannelLayoutTag_Mono) {
if (pos)
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
*channels = 1;
*channel_mask = 0;
return TRUE;
} else if (layout->mChannelLayoutTag == kAudioChannelLayoutTag_Stereo ||
layout->mChannelLayoutTag == kAudioChannelLayoutTag_StereoHeadphones ||
layout->mChannelLayoutTag == kAudioChannelLayoutTag_Binaural) {
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
*channels = 2;
*channel_mask =
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT);
return TRUE;
} else if (layout->mChannelLayoutTag == kAudioChannelLayoutTag_Quadraphonic) {
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2] = GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT;
pos[3] = GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT;
}
*channels = 4;
*channel_mask =
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT) |
GST_AUDIO_CHANNEL_POSITION_MASK (SURROUND_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (SURROUND_RIGHT);
return TRUE;
} else if (layout->mChannelLayoutTag == kAudioChannelLayoutTag_Pentagonal) {
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2] = GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT;
pos[3] = GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT;
pos[4] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
}
*channels = 5;
*channel_mask =
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT) |
GST_AUDIO_CHANNEL_POSITION_MASK (SURROUND_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (SURROUND_RIGHT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_CENTER);
return TRUE;
} else if (layout->mChannelLayoutTag == kAudioChannelLayoutTag_Cube) {
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
pos[4] = GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT;
pos[5] = GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT;
pos[6] = GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT;
pos[7] = GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT;
}
*channels = 8;
*channel_mask =
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT) |
GST_AUDIO_CHANNEL_POSITION_MASK (REAR_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (REAR_RIGHT) |
GST_AUDIO_CHANNEL_POSITION_MASK (TOP_FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (TOP_FRONT_RIGHT) |
GST_AUDIO_CHANNEL_POSITION_MASK (TOP_REAR_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (TOP_REAR_RIGHT);
return TRUE;
} else {
GST_WARNING
("AudioChannelLayoutTag: %u not yet supported",
layout->mChannelLayoutTag);
*channels = 0;
*channel_mask = 0;
return FALSE;
}
}
/* Converts an AudioStreamBasicDescription to preferred caps.
*
* These caps will indicate the AU element's canonical format, which won't
* make Core Audio resample nor convert.
*
* NOTE ON MULTI-CHANNEL AUDIO:
*
* If layout is not NULL, resulting caps will only include the subset
* of channels supported by GStreamer. If the Core Audio layout contained
* ANY positioned channels, then ONLY positioned channels will be included
* in the resulting caps. Otherwise, resulting caps will be unpositioned,
* and include only unpositioned channels.
* (Channels with unsupported AudioChannelLabel will be skipped either way.)
*
* Naturally, the number of channels indicated by 'channels' can be lower
* than the AU element's total number of channels.
*/
GstCaps *
gst_core_audio_asbd_to_caps (AudioStreamBasicDescription * asbd,
AudioChannelLayout * layout)
{
GstAudioInfo info;
GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
guint rate, channels, bps, endianness;
guint64 channel_mask;
gboolean sign;
GstAudioChannelPosition pos[GST_OSX_AUDIO_MAX_CHANNEL];
if (asbd->mFormatID != kAudioFormatLinearPCM) {
GST_WARNING ("Only linear PCM is supported");
goto error;
}
if (!(asbd->mFormatFlags & kAudioFormatFlagIsPacked)) {
GST_WARNING ("Only packed formats supported");
goto error;
}
if (asbd->mFormatFlags & kLinearPCMFormatFlagsSampleFractionMask) {
GST_WARNING ("Fixed point audio is unsupported");
goto error;
}
rate = asbd->mSampleRate;
if (rate == kAudioStreamAnyRate) {
GST_WARNING ("No sample rate");
goto error;
}
bps = asbd->mBitsPerChannel;
endianness = asbd->mFormatFlags & kAudioFormatFlagIsBigEndian ?
G_BIG_ENDIAN : G_LITTLE_ENDIAN;
sign = asbd->mFormatFlags & kAudioFormatFlagIsSignedInteger ? TRUE : FALSE;
if (asbd->mFormatFlags & kAudioFormatFlagIsFloat) {
if (bps == 32) {
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_F32LE;
else
format = GST_AUDIO_FORMAT_F32BE;
} else if (bps == 64) {
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_F64LE;
else
format = GST_AUDIO_FORMAT_F64BE;
}
} else {
format = gst_audio_format_build_integer (sign, endianness, bps, bps);
}
if (format == GST_AUDIO_FORMAT_UNKNOWN) {
GST_WARNING ("Unsupported sample format");
goto error;
}
if (layout) {
if (!gst_core_audio_parse_channel_layout (layout, &channels, &channel_mask,
pos)) {
GST_WARNING
("Failed to parse channel layout, best effort channels layout mapping will be used");
layout = NULL;
}
}
if (layout) {
/* The AU can have arbitrary channel order, but we're using GstAudioInfo
* which supports only the GStreamer channel order.
* Also, we're eventually producing caps, which only have channel-mask
* (whose implied order is the GStreamer channel order). */
gst_audio_channel_positions_to_valid_order (pos, channels);
gst_audio_info_set_format (&info, format, rate, channels, pos);
} else {
channels = MIN (asbd->mChannelsPerFrame, GST_OSX_AUDIO_MAX_CHANNEL);
gst_audio_info_set_format (&info, format, rate, channels, NULL);
}
return gst_audio_info_to_caps (&info);
error:
return NULL;
}
static gboolean
_core_audio_get_property (GstCoreAudio * core_audio, gboolean outer,
AudioUnitPropertyID inID, void *inData, UInt32 * inDataSize)
{
OSStatus status;
AudioUnitScope scope;
AudioUnitElement element;
scope = outer ?
CORE_AUDIO_OUTER_SCOPE (core_audio) : CORE_AUDIO_INNER_SCOPE (core_audio);
element = CORE_AUDIO_ELEMENT (core_audio);
status =
AudioUnitGetProperty (core_audio->audiounit, inID, scope, element, inData,
inDataSize);
return status == noErr;
}
static gboolean
_core_audio_get_stream_format (GstCoreAudio * core_audio,
AudioStreamBasicDescription * asbd, gboolean outer)
{
UInt32 size;
size = sizeof (AudioStreamBasicDescription);
return _core_audio_get_property (core_audio, outer,
kAudioUnitProperty_StreamFormat, asbd, &size);
}
AudioChannelLayout *
gst_core_audio_get_channel_layout (GstCoreAudio * core_audio, gboolean outer)
{
UInt32 size;
AudioChannelLayout *layout;
if (core_audio->is_src) {
GST_WARNING_OBJECT (core_audio,
"gst_core_audio_get_channel_layout not supported on source.");
return NULL;
}
if (!_core_audio_get_property (core_audio, outer,
kAudioUnitProperty_AudioChannelLayout, NULL, &size)) {
GST_WARNING_OBJECT (core_audio, "unable to get channel layout");
return NULL;
}
layout = g_malloc (size);
if (!_core_audio_get_property (core_audio, outer,
kAudioUnitProperty_AudioChannelLayout, layout, &size)) {
GST_WARNING_OBJECT (core_audio, "unable to get channel layout");
g_free (layout);
return NULL;
}
return layout;
}
#define STEREO_CHANNEL_MASK \
(GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) | \
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT))
GstCaps *
gst_core_audio_probe_caps (GstCoreAudio * core_audio, GstCaps * in_caps)
{
guint i, channels, channels_max = 0;
gboolean spdif_allowed;
AudioChannelLayout *layout;
AudioStreamBasicDescription outer_asbd;
gboolean got_outer_asbd;
GstCaps *caps = NULL;
guint64 channel_mask;
/* Get the ASBD of the outer scope (i.e. input scope of Input,
* output scope of Output).
* This ASBD indicates the hardware format. */
got_outer_asbd =
_core_audio_get_stream_format (core_audio, &outer_asbd, TRUE);
/* Collect info about the HW capabilities and preferences */
spdif_allowed =
gst_core_audio_audio_device_is_spdif_avail (core_audio->device_id);
if (!core_audio->is_src)
layout = gst_core_audio_get_channel_layout (core_audio, TRUE);
else
layout = NULL; /* no supported for sources */
GST_DEBUG_OBJECT (core_audio, "Selected device ID: %u SPDIF allowed: %d",
(unsigned) core_audio->device_id, spdif_allowed);
if (layout) {
if (!gst_core_audio_parse_channel_layout (layout, &channels, &channel_mask,
NULL)) {
GST_WARNING_OBJECT (core_audio, "Failed to parse channel layout");
channel_mask = 0;
}
if (channel_mask != 0 && channels > 2 &&
layout->mChannelLayoutTag ==
kAudioChannelLayoutTag_UseChannelDescriptions) {
/* CoreAudio gave us a positioned layout, which might mean we're ignoring some unpositioned channels.
* For example, with a 64ch output, macOS only allows assigning positions to 16 channels at most.
* Let's make sure we also expose the actual maximum amount of channels in our caps,
* without any positions assigned. */
channels_max =
MIN (layout->mNumberChannelDescriptions, GST_OSX_AUDIO_MAX_CHANNEL);
}
/* If available, start with the preferred caps. */
if (got_outer_asbd)
caps = gst_core_audio_asbd_to_caps (&outer_asbd, layout);
g_free (layout);
} else if (got_outer_asbd) {
channels = outer_asbd.mChannelsPerFrame;
channel_mask = 0;
/* If available, start with the preferred caps */
caps = gst_core_audio_asbd_to_caps (&outer_asbd, NULL);
} else {
GST_ERROR_OBJECT (core_audio,
"Unable to get any information about hardware");
return NULL;
}
/* Append the allowed subset based on the template caps */
if (!caps)
caps = gst_caps_new_empty ();
for (i = 0; i < gst_caps_get_size (in_caps); i++) {
GstStructure *in_s;
in_s = gst_caps_get_structure (in_caps, i);
if (gst_structure_has_name (in_s, "audio/x-ac3") ||
gst_structure_has_name (in_s, "audio/x-dts")) {
if (spdif_allowed) {
gst_caps_append_structure (caps, gst_structure_copy (in_s));
}
} else {
GstStructure *out_s;
out_s = gst_structure_copy (in_s);
gst_structure_set (out_s, "channels", G_TYPE_INT, channels, NULL);
if (channel_mask != 0) {
/* positioned layout */
gst_structure_set (out_s,
"channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
} else {
/* unpositioned layout */
gst_structure_remove_field (out_s, "channel-mask");
}
#ifndef HAVE_IOS
if (core_audio->is_src && got_outer_asbd
&& outer_asbd.mSampleRate != kAudioStreamAnyRate) {
/* According to Core Audio engineer, AUHAL does not support sample rate conversion.
* on sources. Therefore, we fixate the sample rate.
*
* "You definitely cannot do rate conversion as part of getting input from AUHAL.
* That's the most common cause of those "cannot do in current context" errors."
* http://lists.apple.com/archives/coreaudio-api/2006/Sep/msg00088.html
*/
gst_structure_set (out_s, "rate", G_TYPE_INT,
(gint) outer_asbd.mSampleRate, NULL);
}
#endif
/* Special cases for upmixing and downmixing.
* Other than that, the AUs don't upmix or downmix multi-channel audio,
* e.g. if you push 5.1-surround audio to a stereo configuration,
* the left and right channels will be played accordingly,
* and the rest will be dropped. */
if (channels == 1) {
/* If have mono, then also offer stereo since CoreAudio downmixes to it */
GstStructure *stereo = gst_structure_copy (out_s);
gst_structure_remove_field (out_s, "channel-mask");
gst_structure_set (stereo, "channels", G_TYPE_INT, 2,
"channel-mask", GST_TYPE_BITMASK, STEREO_CHANNEL_MASK, NULL);
gst_caps_append_structure (caps, stereo);
gst_caps_append_structure (caps, out_s);
} else if (channels == 2 && (channel_mask == 0
|| channel_mask == STEREO_CHANNEL_MASK)) {
/* If have stereo channels, then also offer mono since CoreAudio
* upmixes it. */
GstStructure *mono = gst_structure_copy (out_s);
gst_structure_set (mono, "channels", G_TYPE_INT, 1, NULL);
gst_structure_remove_field (mono, "channel-mask");
gst_structure_set (out_s, "channel-mask", GST_TYPE_BITMASK,
STEREO_CHANNEL_MASK, NULL);
gst_caps_append_structure (caps, out_s);
gst_caps_append_structure (caps, mono);
} else {
/* Otherwise, if needed, add an unpositioned max-channels variant ... */
if (channels_max > 0) {
GstStructure *unpos_s = gst_structure_copy (in_s);
gst_structure_set (unpos_s, "channels", G_TYPE_INT, channels_max,
NULL);
gst_structure_set (unpos_s, "channel-mask", GST_TYPE_BITMASK, 0,
NULL);
gst_caps_append_structure (caps, unpos_s);
}
/* ... and just add the caps */
gst_caps_append_structure (caps, out_s);
}
}
}
GST_DEBUG_OBJECT (core_audio, "Probed caps:%" GST_PTR_FORMAT, caps);
return caps;
}
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