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mpv/audio/out/ao_coreaudio.c

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/*
* CoreAudio audio output driver for Mac OS X
*
* original copyright (C) Timothy J. Wood - Aug 2000
* ported to MPlayer libao2 by Dan Christiansen
*
* Chris Roccati
* Stefano Pigozzi
*
* The S/PDIF part of the code is based on the auhal audio output
* module from VideoLAN:
* Copyright (c) 2006 Derk-Jan Hartman <hartman at videolan dot org>
*
* This file is part of MPlayer.
*
* MPlayer 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.
*
* MPlayer 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
* along with MPlayer; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* The MacOS X CoreAudio framework doesn't mesh as simply as some
* simpler frameworks do. This is due to the fact that CoreAudio pulls
* audio samples rather than having them pushed at it (which is nice
* when you are wanting to do good buffering of audio).
*/
#include "config.h"
#include "ao.h"
#include "audio/format.h"
#include "osdep/timer.h"
#include "options/m_option.h"
#include "misc/ring.h"
#include "common/msg.h"
#include "audio/out/ao_coreaudio_properties.h"
#include "audio/out/ao_coreaudio_utils.h"
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 ao *ao, struct mp_ring *buffer)
{
void *tctx = talloc_new(NULL);
MP_VERBOSE(ao, "%s\n", mp_ring_repr(buffer, tctx));
talloc_free(tctx);
}
struct priv_d {
// digital render callback
AudioDeviceIOProcID render_cb;
// pid set for hog mode, (-1) means that hog mode on the device was
// released. hog mode is exclusive access to a device
pid_t hog_pid;
// stream selected for digital playback by the detection in init
AudioStreamID stream;
// stream index in an AudioBufferList
int stream_idx;
// format we changed the stream to: for the digital case each application
// sets the stream format for a device to what it needs
AudioStreamBasicDescription stream_asbd;
AudioStreamBasicDescription original_asbd;
bool changed_mixing;
int stream_asbd_changed;
bool muted;
};
struct priv {
AudioDeviceID device; // selected device
bool is_digital; // running in digital mode?
AudioUnit audio_unit; // AudioUnit for lpcm output
bool paused;
struct mp_ring *buffer;
struct priv_d *digital;
// options
int opt_device_id;
int opt_list;
};
static int get_ring_size(struct ao *ao)
{
return af_fmt_seconds_to_bytes(
ao->format, 0.5, ao->channels.num, ao->samplerate);
}
static OSStatus render_cb_lpcm(void *ctx, AudioUnitRenderActionFlags *aflags,
const AudioTimeStamp *ts, UInt32 bus,
UInt32 frames, AudioBufferList *buffer_list)
{
struct ao *ao = ctx;
struct priv *p = ao->priv;
AudioBuffer buf = buffer_list->mBuffers[0];
int requested = buf.mDataByteSize;
if (mp_ring_buffered(p->buffer) < requested) {
MP_VERBOSE(ao, "buffer underrun\n");
audio_pause(ao);
memset(buf.mData, 0, requested);
} else {
mp_ring_read(p->buffer, buf.mData, requested);
}
return noErr;
}
static OSStatus render_cb_digital(
AudioDeviceID device, const AudioTimeStamp *ts,
const void *in_data, const AudioTimeStamp *in_ts,
AudioBufferList *out_data, const AudioTimeStamp *out_ts, void *ctx)
{
struct ao *ao = ctx;
struct priv *p = ao->priv;
struct priv_d *d = p->digital;
AudioBuffer buf = out_data->mBuffers[d->stream_idx];
int requested = buf.mDataByteSize;
if (d->muted)
mp_ring_drain(p->buffer, requested);
else
mp_ring_read(p->buffer, buf.mData, requested);
return noErr;
}
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct priv *p = ao->priv;
ao_control_vol_t *control_vol;
OSStatus err;
Float32 vol;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
control_vol = (ao_control_vol_t *)arg;
if (p->is_digital) {
struct priv_d *d = p->digital;
// Digital output has no volume adjust.
int digitalvol = d->muted ? 0 : 100;
*control_vol = (ao_control_vol_t) {
.left = digitalvol, .right = digitalvol,
};
return CONTROL_TRUE;
}
err = AudioUnitGetParameter(p->audio_unit, kHALOutputParam_Volume,
kAudioUnitScope_Global, 0, &vol);
CHECK_CA_ERROR("could not get HAL output volume");
control_vol->left = control_vol->right = vol * 100.0;
return CONTROL_TRUE;
case AOCONTROL_SET_VOLUME:
control_vol = (ao_control_vol_t *)arg;
if (p->is_digital) {
struct priv_d *d = p->digital;
// Digital output can not set volume. Here we have to return true
// to make mixer forget it. Else mixer will add a soft filter,
// that's not we expected and the filter not support ac3 stream
// will cause mplayer die.
// Although not support set volume, but at least we support mute.
// MPlayer set mute by set volume to zero, we handle it.
if (control_vol->left == 0 && control_vol->right == 0)
d->muted = true;
else
d->muted = false;
return CONTROL_TRUE;
}
vol = (control_vol->left + control_vol->right) / 200.0;
err = AudioUnitSetParameter(p->audio_unit, kHALOutputParam_Volume,
kAudioUnitScope_Global, 0, vol, 0);
CHECK_CA_ERROR("could not set HAL output volume");
return CONTROL_TRUE;
} // end switch
return CONTROL_UNKNOWN;
coreaudio_error:
return CONTROL_ERROR;
}
static void print_list(struct ao *ao)
{
char *help = talloc_strdup(NULL, "Available output devices:\n");
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++) {
char *name;
err = CA_GET_STR(devs[i], kAudioObjectPropertyName, &name);
if (err == noErr)
talloc_steal(devs, name);
else
name = "Unknown";
help = talloc_asprintf_append(
help, " * %s (id: %" PRIu32 ")\n", name, devs[i]);
}
talloc_free(devs);
coreaudio_error:
MP_INFO(ao, "%s", help);
talloc_free(help);
}
static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd);
static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd);
static int init(struct ao *ao)
{
OSStatus err;
struct priv *p = ao->priv;
if (p->opt_list) print_list(ao);
struct priv_d *d = talloc_zero(p, struct priv_d);
*d = (struct priv_d) {
.muted = false,
.stream_asbd_changed = 0,
.hog_pid = -1,
.stream = 0,
.stream_idx = -1,
.changed_mixing = false,
};
p->digital = d;
ao->per_application_mixer = true;
ao->no_persistent_volume = true;
AudioDeviceID selected_device = 0;
if (p->opt_device_id < 0) {
// device not set by user, get the default one
err = CA_GET(kAudioObjectSystemObject,
kAudioHardwarePropertyDefaultOutputDevice,
&selected_device);
CHECK_CA_ERROR("could not get default audio device");
} else {
selected_device = p->opt_device_id;
}
if (mp_msg_test(ao->log, MSGL_V)) {
char *name;
err = CA_GET_STR(selected_device, kAudioObjectPropertyName, &name);
CHECK_CA_ERROR("could not get selected audio device name");
MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
name, selected_device);
talloc_free(name);
}
// Save selected device id
p->device = selected_device;
ao->format = af_fmt_from_planar(ao->format);
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(ao, selected_device))
supports_digital = true;
}
if (!supports_digital) {
AudioChannelLayout *layouts;
size_t n_layouts;
err = CA_GET_ARY_O(selected_device,
kAudioDevicePropertyPreferredChannelLayout,
&layouts, &n_layouts);
CHECK_CA_ERROR("could not get audio device prefered layouts");
uint32_t *bitmaps;
size_t n_bitmaps;
ca_bitmaps_from_layouts(ao, layouts, n_layouts, &bitmaps, &n_bitmaps);
talloc_free(layouts);
struct mp_chmap_sel chmap_sel = {0};
for (int i=0; i < n_bitmaps; i++) {
struct mp_chmap chmap = {0};
mp_chmap_from_lavc(&chmap, bitmaps[i]);
mp_chmap_sel_add_map(&chmap_sel, &chmap);
}
talloc_free(bitmaps);
if (ao->channels.num < 3 || n_bitmaps < 1)
// If the input is not surround or we could not get any usable
// bitmap from the hardware, default to waveext...
mp_chmap_sel_add_waveext(&chmap_sel);
if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels))
goto coreaudio_error;
} // closes if (!supports_digital)
// Build ASBD for the input format
AudioStreamBasicDescription asbd;
asbd.mSampleRate = ao->samplerate;
asbd.mFormatID = supports_digital ?
kAudioFormat60958AC3 : kAudioFormatLinearPCM;
asbd.mChannelsPerFrame = ao->channels.num;
asbd.mBitsPerChannel = af_fmt2bits(ao->format);
asbd.mFormatFlags = kAudioFormatFlagIsPacked;
if ((ao->format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F)
asbd.mFormatFlags |= kAudioFormatFlagIsFloat;
if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI)
asbd.mFormatFlags |= kAudioFormatFlagIsSignedInteger;
if ((ao->format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
asbd.mFormatFlags |= kAudioFormatFlagIsBigEndian;
asbd.mFramesPerPacket = 1;
asbd.mBytesPerPacket = asbd.mBytesPerFrame =
asbd.mFramesPerPacket * asbd.mChannelsPerFrame *
(asbd.mBitsPerChannel / 8);
ca_print_asbd(ao, "source format:", &asbd);
if (supports_digital)
return init_digital(ao, asbd);
else
return init_lpcm(ao, asbd);
coreaudio_error:
return CONTROL_ERROR;
}
static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
{
OSStatus err;
uint32_t size;
struct priv *p = ao->priv;
AudioComponentDescription desc = (AudioComponentDescription) {
.componentType = kAudioUnitType_Output,
.componentSubType = (p->opt_device_id < 0) ?
kAudioUnitSubType_DefaultOutput :
kAudioUnitSubType_HALOutput,
.componentManufacturer = kAudioUnitManufacturer_Apple,
.componentFlags = 0,
.componentFlagsMask = 0,
};
AudioComponent comp = AudioComponentFindNext(NULL, &desc);
if (comp == NULL) {
MP_ERR(ao, "unable to find audio component\n");
goto coreaudio_error;
}
err = AudioComponentInstanceNew(comp, &(p->audio_unit));
CHECK_CA_ERROR("unable to open audio component");
// Initialize AudioUnit
err = AudioUnitInitialize(p->audio_unit);
CHECK_CA_ERROR_L(coreaudio_error_component,
"unable to initialize audio unit");
size = sizeof(AudioStreamBasicDescription);
err = AudioUnitSetProperty(p->audio_unit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &asbd, size);
CHECK_CA_ERROR_L(coreaudio_error_audiounit,
"unable to set the input format on the audio unit");
//Set the Current Device to the Default Output Unit.
err = AudioUnitSetProperty(p->audio_unit,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global, 0, &p->device,
sizeof(p->device));
CHECK_CA_ERROR_L(coreaudio_error_audiounit,
"can't link audio unit to selected device");
if (ao->channels.num > 2) {
// No need to set a channel layout for mono and stereo inputs
AudioChannelLayout acl = (AudioChannelLayout) {
.mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelBitmap,
.mChannelBitmap = mp_chmap_to_waveext(&ao->channels)
};
err = AudioUnitSetProperty(p->audio_unit,
kAudioUnitProperty_AudioChannelLayout,
kAudioUnitScope_Input, 0, &acl,
sizeof(AudioChannelLayout));
CHECK_CA_ERROR_L(coreaudio_error_audiounit,
"can't set channel layout bitmap into audio unit");
}
p->buffer = mp_ring_new(p, get_ring_size(ao));
print_buffer(ao, p->buffer);
AURenderCallbackStruct render_cb = (AURenderCallbackStruct) {
.inputProc = render_cb_lpcm,
.inputProcRefCon = ao,
};
err = AudioUnitSetProperty(p->audio_unit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &render_cb,
sizeof(AURenderCallbackStruct));
CHECK_CA_ERROR_L(coreaudio_error_audiounit,
"unable to set render callback on audio unit");
reset(ao);
return CONTROL_OK;
coreaudio_error_audiounit:
AudioUnitUninitialize(p->audio_unit);
coreaudio_error_component:
AudioComponentInstanceDispose(p->audio_unit);
coreaudio_error:
return CONTROL_ERROR;
}
static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
{
struct priv *p = ao->priv;
struct priv_d *d = p->digital;
OSStatus err = noErr;
uint32_t is_alive = 1;
err = CA_GET(p->device, kAudioDevicePropertyDeviceIsAlive, &is_alive);
CHECK_CA_WARN("could not check whether device is alive");
if (!is_alive)
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(ao, p->device, &d->changed_mixing);
CHECK_CA_WARN("failed to disable mixing");
AudioStreamID *streams;
size_t n_streams;
/* Get a list of all the streams on this device. */
err = CA_GET_ARY_O(p->device, kAudioDevicePropertyStreams,
&streams, &n_streams);
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(ao, streams[i]);
if (digital) {
err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat,
&d->original_asbd);
if (!CHECK_CA_WARN("could not get stream's physical format to "
"revert to, getting the next one"))
continue;
AudioStreamRangedDescription *formats;
size_t n_formats;
err = CA_GET_ARY(streams[i],
kAudioStreamPropertyAvailablePhysicalFormats,
&formats, &n_formats);
if (!CHECK_CA_WARN("could not get number of stream formats"))
continue; // try next one
int req_rate_format = -1;
int max_rate_format = -1;
d->stream = streams[i];
d->stream_idx = i;
for (int j = 0; j < n_formats; j++)
if (ca_format_is_digital(formats[j].mFormat)) {
// select the digital format that has exactly the same
// samplerate. If an exact match cannot be found, select
// the format with highest samplerate as backup.
if (formats[j].mFormat.mSampleRate == asbd.mSampleRate) {
req_rate_format = j;
break;
} else if (max_rate_format < 0 ||
formats[j].mFormat.mSampleRate >
formats[max_rate_format].mFormat.mSampleRate)
max_rate_format = j;
}
if (req_rate_format >= 0)
d->stream_asbd = formats[req_rate_format].mFormat;
else
d->stream_asbd = formats[max_rate_format].mFormat;
talloc_free(formats);
}
}
talloc_free(streams);
if (d->stream_idx < 0) {
MP_WARN(ao , "can't find any digital output stream format\n");
goto coreaudio_error;
}
if (!ca_change_format(ao, d->stream, d->stream_asbd))
goto coreaudio_error;
void *changed = (void *) &(d->stream_asbd_changed);
err = ca_enable_device_listener(p->device, changed);
CHECK_CA_ERROR("cannot install format change listener during init");
#if BYTE_ORDER == BIG_ENDIAN
if (!(p->stream_asdb.mFormatFlags & kAudioFormatFlagIsBigEndian))
#else
/* tell mplayer that we need a byteswap on AC3 streams, */
if (d->stream_asbd.mFormatID & kAudioFormat60958AC3)
ao->format = AF_FORMAT_AC3_LE;
else if (d->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
MP_WARN(ao, "stream has non-native byte order, output may fail\n");
ao->samplerate = d->stream_asbd.mSampleRate;
ao->bps = ao->samplerate *
(d->stream_asbd.mBytesPerPacket /
d->stream_asbd.mFramesPerPacket);
p->buffer = mp_ring_new(p, get_ring_size(ao));
print_buffer(ao, p->buffer);
err = AudioDeviceCreateIOProcID(p->device,
(AudioDeviceIOProc)render_cb_digital,
(void *)ao,
&d->render_cb);
CHECK_CA_ERROR("failed to register digital render callback");
reset(ao);
return CONTROL_TRUE;
coreaudio_error:
err = ca_unlock_device(p->device, &d->hog_pid);
CHECK_CA_WARN("can't release hog mode");
return CONTROL_ERROR;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *p = ao->priv;
struct priv_d *d = p->digital;
void *output_samples = data[0];
int num_bytes = samples * ao->sstride;
// 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(ao, d->stream)) {
if (!ca_change_format(ao, d->stream, d->stream_asbd)) {
MP_WARN(ao , "can't restore digital output\n");
} else {
MP_WARN(ao, "restoring digital output succeeded.\n");
reset(ao);
}
}
}
int wrote = mp_ring_write(p->buffer, output_samples, num_bytes);
audio_resume(ao);
return wrote / ao->sstride;
}
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
audio_pause(ao);
mp_ring_reset(p->buffer);
}
static int get_space(struct ao *ao)
{
struct priv *p = ao->priv;
return mp_ring_available(p->buffer) / ao->sstride;
}
static float get_delay(struct ao *ao)
{
// FIXME: should also report the delay of coreaudio itself (hardware +
// internal buffers)
struct priv *p = ao->priv;
return mp_ring_buffered(p->buffer) / (float)ao->bps;
}
static void uninit(struct ao *ao, bool immed)
{
struct priv *p = ao->priv;
OSStatus err = noErr;
if (!immed)
mp_sleep_us(get_delay(ao) * 1000000);
if (!p->is_digital) {
AudioOutputUnitStop(p->audio_unit);
AudioUnitUninitialize(p->audio_unit);
AudioComponentInstanceDispose(p->audio_unit);
} else {
struct priv_d *d = p->digital;
void *changed = (void *) &(d->stream_asbd_changed);
err = ca_disable_device_listener(p->device, changed);
CHECK_CA_WARN("can't remove device listener, this may cause a crash");
err = AudioDeviceStop(p->device, d->render_cb);
CHECK_CA_WARN("failed to stop audio device");
err = AudioDeviceDestroyIOProcID(p->device, d->render_cb);
CHECK_CA_WARN("failed to remove device render callback");
if (!ca_change_format(ao, d->stream, d->original_asbd))
MP_WARN(ao, "can't revert to original device format");
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);
CHECK_CA_WARN("can't release hog mode");
}
}
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
OSErr err = noErr;
if (p->paused)
return;
if (!p->is_digital) {
err = AudioOutputUnitStop(p->audio_unit);
CHECK_CA_WARN("can't stop audio unit");
} else {
struct priv_d *d = p->digital;
err = AudioDeviceStop(p->device, d->render_cb);
CHECK_CA_WARN("can't stop digital device");
}
p->paused = true;
}
static void audio_resume(struct ao *ao)
{
struct priv *p = ao->priv;
OSErr err = noErr;
if (!p->paused)
return;
if (!p->is_digital) {
err = AudioOutputUnitStart(p->audio_unit);
CHECK_CA_WARN("can't start audio unit");
} else {
struct priv_d *d = p->digital;
err = AudioDeviceStart(p->device, d->render_cb);
CHECK_CA_WARN("can't start digital device");
}
p->paused = false;
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_coreaudio = {
.description = "CoreAudio (OS X Audio Output)",
.name = "coreaudio",
.uninit = uninit,
.init = init,
.play = play,
.control = control,
.get_space = get_space,
.get_delay = get_delay,
.reset = reset,
.pause = audio_pause,
.resume = audio_resume,
.priv_size = sizeof(struct priv),
.options = (const struct m_option[]) {
OPT_INT("device_id", opt_device_id, 0, OPTDEF_INT(-1)),
OPT_FLAG("list", opt_list, 0),
{0}
},
};
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