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f9a31bc3d9
mpv/audio/out/ao_coreaudio.c
Stefano Pigozzi f9a31bc3d9 ao_coreaudio: refactor print_help
2013-07-22 21:53:17 +02:00

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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
*
* 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 "audio/out/ao_coreaudio_common.c"
#include "ao.h"
#include "audio/format.h"
#include "osdep/timer.h"
#include "core/subopt-helper.h"
#include "core/mp_ring.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 mp_ring *buffer)
{
void *tctx = talloc_new(NULL);
ca_msg(MSGL_V, "%s\n", mp_ring_repr(buffer, tctx));
talloc_free(tctx);
}
struct priv
{
AudioDeviceID i_selected_dev; /* Keeps DeviceID of the selected device. */
int b_supports_digital; /* Does the currently selected device support digital mode? */
int b_digital; /* Are we running in digital mode? */
int b_muted; /* Are we muted in digital mode? */
AudioDeviceIOProcID renderCallback; /* Render callback used for SPDIF */
/* AudioUnit */
AudioUnit theOutputUnit;
/* CoreAudio SPDIF mode specific */
pid_t i_hog_pid; /* Keeps the pid of our hog status. */
AudioStreamID i_stream_id; /* The StreamID that has a cac3 streamformat */
int i_stream_index; /* The index of i_stream_id in an AudioBufferList */
AudioStreamBasicDescription stream_format; /* The format we changed the stream to */
AudioStreamBasicDescription sfmt_revert; /* The original format of the stream */
int b_revert; /* Whether we need to revert the stream format */
int b_changed_mixing; /* Whether we need to set the mixing mode back */
int b_stream_format_changed; /* Flag for main thread to reset stream's format to digital and reset buffer */
/* Original common part */
int packetSize;
int paused;
struct mp_ring *buffer;
};
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;
int requested = frames * p->packetSize;
AudioBuffer buf = buffer_list->mBuffers[0];
buf.mDataByteSize = 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;
AudioBuffer buf = out_data->mBuffers[p->i_stream_index];
int requested = buf.mDataByteSize;
if (p->b_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->b_digital) {
// Digital output has no volume adjust.
int vol = p->b_muted ? 0 : 100;
*control_vol = (ao_control_vol_t) {
.left = vol, .right = vol,
};
return CONTROL_TRUE;
}
err = AudioUnitGetParameter(p->theOutputUnit, kHALOutputParam_Volume,
kAudioUnitScope_Global, 0, &vol);
CHECK_CA_ERROR("could not get HAL output volume");
control_vol->left = control_vol->right = vol * 100.0 / 4.0;
return CONTROL_TRUE;
case AOCONTROL_SET_VOLUME:
control_vol = (ao_control_vol_t *)arg;
if (p->b_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)
p->b_muted = 1;
else
p->b_muted = 0;
return CONTROL_TRUE;
}
vol = (control_vol->left + control_vol->right) * 4.0 / 200.0;
err = AudioUnitSetParameter(p->theOutputUnit, 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 int OpenSPDIF(struct ao *ao);
static int AudioStreamChangeFormat(AudioStreamID i_stream_id,
AudioStreamBasicDescription change_format);
static void print_help(void)
{
ca_msg(MSGL_FATAL,
"\n-ao coreaudio commandline help:\n"
"Example: mpv -ao coreaudio:device_id=266\n"
" open Core Audio with output device ID 266.\n"
"\nOptions:\n"
" device_id\n"
" ID of output device to use (0 = default device)\n"
" help\n"
" This help including list of available devices.\n"
"\n"
"Available output devices:\n");
AudioDeviceID *devs;
uint32_t devs_size =
GetGlobalAudioPropertyArray(kAudioObjectSystemObject,
kAudioHardwarePropertyDevices,
(void **)&devs);
if (!devs_size) {
ca_msg(MSGL_FATAL, "Failed to get list of output devices.\n");
return;
}
int devs_n = devs_size / sizeof(AudioDeviceID);
for (int i = 0; i < devs_n; ++i) {
char *name;
OSStatus err =
GetAudioPropertyString(devs[i], kAudioObjectPropertyName, &name);
if (err == noErr) {
ca_msg(MSGL_FATAL, "%s (id: %" PRIu32 ")\n", name, devs[i]);
free(name);
} else
ca_msg(MSGL_FATAL, "Unknown (id: %" PRIu32 ")\n", devs[i]);
}
free(devs);
}
static int init(struct ao *ao, char *params)
{
// int rate, int channels, int format, int flags)
struct priv *p = talloc_zero(ao, struct priv);
ao->priv = p;
AudioStreamBasicDescription inDesc;
AudioComponentDescription desc;
AudioComponent comp;
AURenderCallbackStruct renderCallback;
OSStatus err;
UInt32 size, maxFrames, b_alive;
char *psz_name;
AudioDeviceID devid_def = 0;
int device_id = 0, display_help = 0;
const opt_t subopts[] = {
{"device_id", OPT_ARG_INT, &device_id, NULL},
{"help", OPT_ARG_BOOL, &display_help, NULL},
{NULL}
};
if (subopt_parse(params, subopts) != 0) {
print_help();
return 0;
}
if (display_help)
print_help();
ca_msg(MSGL_V, "init([%dHz][%dch][%s][%d])\n",
ao->samplerate, ao->channels.num, af_fmt2str_short(ao->format), 0);
p->i_selected_dev = 0;
p->b_supports_digital = 0;
p->b_digital = 0;
p->b_muted = 0;
p->b_stream_format_changed = 0;
p->i_hog_pid = -1;
p->i_stream_id = 0;
p->i_stream_index = -1;
p->b_revert = 0;
p->b_changed_mixing = 0;
ao->per_application_mixer = true;
ao->no_persistent_volume = true;
if (device_id == 0) {
/* Find the ID of the default Device. */
err = GetAudioProperty(kAudioObjectSystemObject,
kAudioHardwarePropertyDefaultOutputDevice,
sizeof(UInt32), &devid_def);
if (err != noErr) {
ca_msg(MSGL_WARN,
"could not get default audio device: [%4.4s]\n",
(char *)&err);
goto err_out;
}
} else {
devid_def = device_id;
}
/* Retrieve the name of the device. */
err = GetAudioPropertyString(devid_def,
kAudioObjectPropertyName,
&psz_name);
if (err != noErr) {
ca_msg(MSGL_WARN,
"could not get default audio device name: [%4.4s]\n",
(char *)&err);
goto err_out;
}
ca_msg(MSGL_V,
"got audio output device ID: %" PRIu32 " Name: %s\n", devid_def,
psz_name);
/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
if (AF_FORMAT_IS_AC3(ao->format)) {
if (AudioDeviceSupportsDigital(devid_def))
p->b_supports_digital = 1;
ca_msg(MSGL_V,
"probe default audio output device about support for digital s/pdif output: %d\n",
p->b_supports_digital);
}
free(psz_name);
// Save selected device id
p->i_selected_dev = devid_def;
struct mp_chmap_sel chmap_sel = {0};
mp_chmap_sel_add_waveext(&chmap_sel);
if (!ao_chmap_sel_adjust(ao, &chmap_sel, &ao->channels))
goto err_out;
// Build Description for the input format
inDesc.mSampleRate = ao->samplerate;
inDesc.mFormatID =
p->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
inDesc.mChannelsPerFrame = ao->channels.num;
inDesc.mBitsPerChannel = af_fmt2bits(ao->format);
if ((ao->format & AF_FORMAT_POINT_MASK) == AF_FORMAT_F) {
// float
inDesc.mFormatFlags = kAudioFormatFlagIsFloat |
kAudioFormatFlagIsPacked;
} else if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI) {
// signed int
inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger |
kAudioFormatFlagIsPacked;
} else {
// unsigned int
inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
}
if ((ao->format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
inDesc.mFramesPerPacket = 1;
p->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame =
inDesc.mFramesPerPacket *
ao->channels.num *
(inDesc.mBitsPerChannel / 8);
ca_print_asbd("source format:", &inDesc);
if (p->b_supports_digital) {
b_alive = 1;
err = GetAudioProperty(p->i_selected_dev,
kAudioDevicePropertyDeviceIsAlive,
sizeof(UInt32), &b_alive);
if (err != noErr)
ca_msg(MSGL_WARN,
"could not check whether device is alive: [%4.4s]\n",
(char *)&err);
if (!b_alive)
ca_msg(MSGL_WARN, "device is not alive\n");
/* S/PDIF output need device in HogMode. */
err = GetAudioProperty(p->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(pid_t), &p->i_hog_pid);
if (err != noErr) {
/* This is not a fatal error. Some drivers simply don't support this property. */
ca_msg(MSGL_WARN,
"could not check whether device is hogged: [%4.4s]\n",
(char *)&err);
p->i_hog_pid = -1;
}
if (p->i_hog_pid != -1 && p->i_hog_pid != getpid()) {
ca_msg(MSGL_WARN,
"Selected audio device is exclusively in use by another program.\n");
goto err_out;
}
p->stream_format = inDesc;
return OpenSPDIF(ao);
}
/* original analog output code */
desc.componentType = kAudioUnitType_Output;
desc.componentSubType =
(device_id ==
0) ? kAudioUnitSubType_DefaultOutput : kAudioUnitSubType_HALOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
comp = AudioComponentFindNext(NULL, &desc); //Finds an component that meets the desc spec's
if (comp == NULL) {
ca_msg(MSGL_WARN, "Unable to find Output Unit component\n");
goto err_out;
}
err = AudioComponentInstanceNew(comp, &(p->theOutputUnit)); //gains access to the services provided by the component
if (err) {
ca_msg(MSGL_WARN,
"Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
goto err_out;
}
// Initialize AudioUnit
err = AudioUnitInitialize(p->theOutputUnit);
if (err) {
ca_msg(MSGL_WARN,
"Unable to initialize Output Unit component: [%4.4s]\n",
(char *)&err);
goto err_out1;
}
size = sizeof(AudioStreamBasicDescription);
err = AudioUnitSetProperty(p->theOutputUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &inDesc, size);
if (err) {
ca_msg(MSGL_WARN, "Unable to set the input format: [%4.4s]\n",
(char *)&err);
goto err_out2;
}
size = sizeof(UInt32);
err = AudioUnitGetProperty(p->theOutputUnit,
kAudioDevicePropertyBufferSize,
kAudioUnitScope_Input, 0, &maxFrames, &size);
if (err) {
ca_msg(MSGL_WARN,
"AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n",
(char *)&err);
goto err_out2;
}
//Set the Current Device to the Default Output Unit.
err = AudioUnitSetProperty(p->theOutputUnit,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global, 0, &p->i_selected_dev,
sizeof(p->i_selected_dev));
ao->samplerate = inDesc.mSampleRate;
if (!ao_chmap_sel_get_def(ao, &chmap_sel, &ao->channels,
inDesc.mChannelsPerFrame))
goto err_out2;
ao->bps = ao->samplerate * inDesc.mBytesPerFrame;
p->buffer = mp_ring_new(p, get_ring_size(ao));
print_buffer(p->buffer);
renderCallback.inputProc = render_cb_lpcm;
renderCallback.inputProcRefCon = ao;
err = AudioUnitSetProperty(p->theOutputUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &renderCallback,
sizeof(AURenderCallbackStruct));
if (err) {
ca_msg(MSGL_WARN,
"Unable to set the render callback: [%4.4s]\n", (char *)&err);
goto err_out2;
}
reset(ao);
return CONTROL_OK;
err_out2:
AudioUnitUninitialize(p->theOutputUnit);
err_out1:
AudioComponentInstanceDispose(p->theOutputUnit);
err_out:
return CONTROL_FALSE;
}
/*****************************************************************************
* Setup a encoded digital stream (SPDIF)
*****************************************************************************/
static int OpenSPDIF(struct ao *ao)
{
struct priv *p = ao->priv;
OSStatus err = noErr;
UInt32 i_param_size, b_mix = 0;
Boolean b_writeable = 0;
AudioStreamID *p_streams = NULL;
int i, i_streams = 0;
AudioObjectPropertyAddress p_addr;
/* Start doing the SPDIF setup process. */
p->b_digital = 1;
/* Hog the device. */
p->i_hog_pid = getpid();
err = SetAudioProperty(p->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(p->i_hog_pid), &p->i_hog_pid);
if (err != noErr) {
ca_msg(MSGL_WARN, "failed to set hogmode: [%4.4s]\n",
(char *)&err);
p->i_hog_pid = -1;
goto err_out;
}
p_addr.mSelector = kAudioDevicePropertySupportsMixing;
p_addr.mScope = kAudioObjectPropertyScopeGlobal;
p_addr.mElement = kAudioObjectPropertyElementMaster;
/* Set mixable to false if we are allowed to. */
if (AudioObjectHasProperty(p->i_selected_dev, &p_addr)) {
/* Set mixable to false if we are allowed to. */
err = IsAudioPropertySettable(p->i_selected_dev,
kAudioDevicePropertySupportsMixing,
&b_writeable);
err = GetAudioProperty(p->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
if (err == noErr && b_writeable) {
b_mix = 0;
err = SetAudioProperty(p->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
p->b_changed_mixing = 1;
}
if (err != noErr) {
ca_msg(MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
(char *)&err);
goto err_out;
}
}
/* Get a list of all the streams on this device. */
i_param_size = GetAudioPropertyArray(p->i_selected_dev,
kAudioDevicePropertyStreams,
kAudioDevicePropertyScopeOutput,
(void **)&p_streams);
if (!i_param_size) {
ca_msg(MSGL_WARN, "could not get number of streams.\n");
goto err_out;
}
i_streams = i_param_size / sizeof(AudioStreamID);
ca_msg(MSGL_V, "current device stream number: %d\n", i_streams);
for (i = 0; i < i_streams && p->i_stream_index < 0; ++i) {
/* Find a stream with a cac3 stream. */
AudioStreamRangedDescription *p_format_list = NULL;
int i_formats = 0, j = 0, b_digital = 0;
i_param_size = GetGlobalAudioPropertyArray(p_streams[i],
kAudioStreamPropertyAvailablePhysicalFormats,
(void **)&p_format_list);
if (!i_param_size) {
ca_msg(MSGL_WARN,
"Could not get number of stream formats.\n");
continue;
}
i_formats = i_param_size / sizeof(AudioStreamRangedDescription);
/* Check if one of the supported formats is a digital format. */
for (j = 0; j < i_formats; ++j) {
if (p_format_list[j].mFormat.mFormatID == 'IAC3' ||
p_format_list[j].mFormat.mFormatID == 'iac3' ||
p_format_list[j].mFormat.mFormatID == kAudioFormat60958AC3 ||
p_format_list[j].mFormat.mFormatID == kAudioFormatAC3) {
b_digital = 1;
break;
}
}
if (b_digital) {
/* If this stream supports a digital (cac3) format, then set it. */
int i_requested_rate_format = -1;
int i_current_rate_format = -1;
int i_backup_rate_format = -1;
p->i_stream_id = p_streams[i];
p->i_stream_index = i;
if (p->b_revert == 0) {
/* Retrieve the original format of this stream first if not done so already. */
err = GetAudioProperty(p->i_stream_id,
kAudioStreamPropertyPhysicalFormat,
sizeof(p->sfmt_revert),
&p->sfmt_revert);
if (err != noErr) {
ca_msg(MSGL_WARN,
"Could not retrieve the original stream format: [%4.4s]\n",
(char *)&err);
free(p_format_list);
continue;
}
p->b_revert = 1;
}
for (j = 0; j < i_formats; ++j)
if (p_format_list[j].mFormat.mFormatID == 'IAC3' ||
p_format_list[j].mFormat.mFormatID == 'iac3' ||
p_format_list[j].mFormat.mFormatID ==
kAudioFormat60958AC3 ||
p_format_list[j].mFormat.mFormatID == kAudioFormatAC3) {
if (p_format_list[j].mFormat.mSampleRate ==
p->stream_format.mSampleRate) {
i_requested_rate_format = j;
break;
}
if (p_format_list[j].mFormat.mSampleRate ==
p->sfmt_revert.mSampleRate)
i_current_rate_format = j;
else if (i_backup_rate_format < 0 ||
p_format_list[j].mFormat.mSampleRate >
p_format_list[i_backup_rate_format].mFormat.
mSampleRate)
i_backup_rate_format = j;
}
if (i_requested_rate_format >= 0) /* We prefer to output at the samplerate of the original audio. */
p->stream_format =
p_format_list[i_requested_rate_format].mFormat;
else if (i_current_rate_format >= 0) /* If not possible, we will try to use the current samplerate of the device. */
p->stream_format =
p_format_list[i_current_rate_format].mFormat;
else
p->stream_format = p_format_list[i_backup_rate_format].mFormat;
/* And if we have to, any digital format will be just fine (highest rate possible). */
}
free(p_format_list);
}
free(p_streams);
if (p->i_stream_index < 0) {
ca_msg(MSGL_WARN,
"Cannot find any digital output stream format when OpenSPDIF().\n");
goto err_out;
}
ca_print_asbd("original stream format:", &p->sfmt_revert);
if (!AudioStreamChangeFormat(p->i_stream_id, p->stream_format))
goto err_out;
p_addr.mSelector = kAudioDevicePropertyDeviceHasChanged;
p_addr.mScope = kAudioObjectPropertyScopeGlobal;
p_addr.mElement = kAudioObjectPropertyElementMaster;
const int *stream_format_changed = &(p->b_stream_format_changed);
err = AudioObjectAddPropertyListener(p->i_selected_dev,
&p_addr,
ca_device_listener,
(void *)stream_format_changed);
if (err != noErr)
ca_msg(MSGL_WARN,
"AudioDeviceAddPropertyListener for kAudioDevicePropertyDeviceHasChanged failed: [%4.4s]\n",
(char *)&err);
/* FIXME: If output stream is not native byte-order, we need change endian somewhere. */
/* Although there's no such case reported. */
#if BYTE_ORDER == BIG_ENDIAN
if (!(p->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
#else
/* tell mplayer that we need a byteswap on AC3 streams, */
if (p->stream_format.mFormatID & kAudioFormat60958AC3)
ao->format = AF_FORMAT_AC3_LE;
if (p->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
ca_msg(MSGL_WARN,
"Output stream has non-native byte order, digital output may fail.\n");
ao->samplerate = p->stream_format.mSampleRate;
mp_chmap_from_channels(&ao->channels, p->stream_format.mChannelsPerFrame);
ao->bps = ao->samplerate *
(p->stream_format.mBytesPerPacket /
p->stream_format.mFramesPerPacket);
p->buffer = mp_ring_new(p, get_ring_size(ao));
print_buffer(p->buffer);
/* Create IOProc callback. */
err = AudioDeviceCreateIOProcID(p->i_selected_dev,
(AudioDeviceIOProc)render_cb_digital,
(void *)ao,
&p->renderCallback);
if (err != noErr || p->renderCallback == NULL) {
ca_msg(MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n",
(char *)&err);
goto err_out1;
}
reset(ao);
return CONTROL_TRUE;
err_out1:
if (p->b_revert)
AudioStreamChangeFormat(p->i_stream_id, p->sfmt_revert);
err_out:
if (p->b_changed_mixing && p->sfmt_revert.mFormatID !=
kAudioFormat60958AC3) {
int b_mix = 1;
err = SetAudioProperty(p->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(int), &b_mix);
if (err != noErr)
ca_msg(MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
(char *)&err);
}
if (p->i_hog_pid == getpid()) {
p->i_hog_pid = -1;
err = SetAudioProperty(p->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(p->i_hog_pid), &p->i_hog_pid);
if (err != noErr)
ca_msg(MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
(char *)&err);
}
return CONTROL_FALSE;
}
/*****************************************************************************
* AudioStreamChangeFormat: Change i_stream_id to change_format
*****************************************************************************/
static int AudioStreamChangeFormat(AudioStreamID i_stream_id,
AudioStreamBasicDescription change_format)
{
OSStatus err = noErr;
int i;
AudioObjectPropertyAddress p_addr;
static volatile int stream_format_changed;
stream_format_changed = 0;
ca_print_asbd("setting stream format:", &change_format);
/* Install the callback. */
p_addr.mSelector = kAudioStreamPropertyPhysicalFormat;
p_addr.mScope = kAudioObjectPropertyScopeGlobal;
p_addr.mElement = kAudioObjectPropertyElementMaster;
err = AudioObjectAddPropertyListener(i_stream_id,
&p_addr,
ca_stream_listener,
(void *)&stream_format_changed);
if (err != noErr) {
ca_msg(MSGL_WARN,
"AudioStreamAddPropertyListener failed: [%4.4s]\n",
(char *)&err);
return CONTROL_FALSE;
}
/* Change the format. */
err = SetAudioProperty(i_stream_id,
kAudioStreamPropertyPhysicalFormat,
sizeof(AudioStreamBasicDescription), &change_format);
if (err != noErr) {
ca_msg(MSGL_WARN, "could not set the stream format: [%4.4s]\n",
(char *)&err);
return CONTROL_FALSE;
}
/* The AudioStreamSetProperty is not only asynchronious,
* it is also not Atomic, in its behaviour.
* Therefore we check 5 times before we really give up.
* FIXME: failing isn't actually implemented yet. */
for (i = 0; i < 5; ++i) {
AudioStreamBasicDescription actual_format;
int j;
for (j = 0; !stream_format_changed && j < 50; ++j)
mp_sleep_us(10000);
if (stream_format_changed)
stream_format_changed = 0;
else
ca_msg(MSGL_V, "reached timeout\n");
err = GetAudioProperty(i_stream_id,
kAudioStreamPropertyPhysicalFormat,
sizeof(AudioStreamBasicDescription),
&actual_format);
ca_print_asbd("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) {
/* The right format is now active. */
break;
}
/* We need to check again. */
}
/* Removing the property listener. */
err = AudioObjectRemovePropertyListener(i_stream_id,
&p_addr,
ca_stream_listener,
(void *)&stream_format_changed);
if (err != noErr) {
ca_msg(MSGL_WARN,
"AudioStreamRemovePropertyListener failed: [%4.4s]\n",
(char *)&err);
return CONTROL_FALSE;
}
return CONTROL_TRUE;
}
static int play(struct ao *ao, void *output_samples, int num_bytes, int flags)
{
struct priv *p = ao->priv;
int wrote, b_digital;
// Check whether we need to reset the digital output stream.
if (p->b_digital && p->b_stream_format_changed) {
p->b_stream_format_changed = 0;
b_digital = AudioStreamSupportsDigital(p->i_stream_id);
if (b_digital) {
/* Current stream supports digital format output, let's set it. */
ca_msg(MSGL_V,
"Detected current stream supports digital, try to restore digital output...\n");
if (!AudioStreamChangeFormat(p->i_stream_id, p->stream_format))
ca_msg(MSGL_WARN,
"Restoring digital output failed.\n");
else {
ca_msg(MSGL_WARN,
"Restoring digital output succeeded.\n");
reset(ao);
}
} else
ca_msg(MSGL_V,
"Detected current stream does not support digital.\n");
}
wrote = mp_ring_write(p->buffer, output_samples, num_bytes);
audio_resume(ao);
return wrote;
}
/* set variables and buffer to initial state */
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
audio_pause(ao);
mp_ring_reset(p->buffer);
}
/* return available space */
static int get_space(struct ao *ao)
{
struct priv *p = ao->priv;
return mp_ring_available(p->buffer);
}
/* return delay until audio is played */
static float get_delay(struct ao *ao)
{
// inaccurate, should also contain the data buffered e.g. by the OS
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) {
long long timeleft =
(1000000LL * mp_ring_buffered(p->buffer)) / ao->bps;
ca_msg(MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n",
mp_ring_buffered(p->buffer), ao->bps, (int)timeleft);
mp_sleep_us((int)timeleft);
}
if (!p->b_digital) {
AudioOutputUnitStop(p->theOutputUnit);
AudioUnitUninitialize(p->theOutputUnit);
AudioComponentInstanceDispose(p->theOutputUnit);
} else {
/* Stop device. */
err = AudioDeviceStop(p->i_selected_dev, p->renderCallback);
if (err != noErr)
ca_msg(MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n",
(char *)&err);
/* Remove IOProc callback. */
err =
AudioDeviceDestroyIOProcID(p->i_selected_dev, p->renderCallback);
if (err != noErr)
ca_msg(MSGL_WARN,
"AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err);
if (p->b_revert)
AudioStreamChangeFormat(p->i_stream_id, p->sfmt_revert);
if (p->b_changed_mixing && p->sfmt_revert.mFormatID !=
kAudioFormat60958AC3) {
UInt32 b_mix;
Boolean b_writeable = 0;
/* Revert mixable to true if we are allowed to. */
err = IsAudioPropertySettable(p->i_selected_dev,
kAudioDevicePropertySupportsMixing,
&b_writeable);
err = GetAudioProperty(p->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
if (err == noErr && b_writeable) {
b_mix = 1;
err = SetAudioProperty(p->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
}
if (err != noErr)
ca_msg(MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
(char *)&err);
}
if (p->i_hog_pid == getpid()) {
p->i_hog_pid = -1;
err = SetAudioProperty(p->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(p->i_hog_pid), &p->i_hog_pid);
if (err != noErr)
ca_msg(MSGL_WARN,
"Could not release hogmode: [%4.4s]\n", (char *)&err);
}
}
}
/* stop playing, keep buffers (for pause) */
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
OSErr err = noErr;
/* Stop callback. */
if (!p->b_digital) {
err = AudioOutputUnitStop(p->theOutputUnit);
if (err != noErr)
ca_msg(MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n",
(char *)&err);
} else {
err = AudioDeviceStop(p->i_selected_dev, p->renderCallback);
if (err != noErr)
ca_msg(MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n",
(char *)&err);
}
p->paused = 1;
}
/* resume playing, after audio_pause() */
static void audio_resume(struct ao *ao)
{
struct priv *p = ao->priv;
OSErr err = noErr;
if (!p->paused)
return;
/* Start callback. */
if (!p->b_digital) {
err = AudioOutputUnitStart(p->theOutputUnit);
if (err != noErr)
ca_msg(MSGL_WARN,
"AudioOutputUnitStart returned [%4.4s]\n", (char *)&err);
} else {
err = AudioDeviceStart(p->i_selected_dev, p->renderCallback);
if (err != noErr)
ca_msg(MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n",
(char *)&err);
}
p->paused = 0;
}
const struct ao_driver audio_out_coreaudio = {
.info = &(const struct ao_info) {
"CoreAudio (Native OS X Audio Output)",
"coreaudio",
"Timothy J. Wood, Dan Christiansen, Chris Roccati & Stefano Pigozzi",
"",
},
.uninit = uninit,
.init = init,
.play = play,
.control = control,
.get_space = get_space,
.get_delay = get_delay,
.reset = reset,
.pause = audio_pause,
.resume = audio_resume,
};
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