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mpv/audio/out/ao_coreaudio.c
Mad Fish 5b7327920b ao_coreaudio: use 0 as timeout for CFRunLoopRunInMode 
Handle all pending events and exit instead of waiting. When there are lots of
input events (for example, scrolling with trackpad), timeout can add up
to make a huge frame delay. In my tests, if I scroll fast enough, that loop
would never exit.
2013-01-20 16:37:30 +01:00

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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).
*
* AC-3 and MPEG audio passthrough is possible, but has never been tested
* due to lack of a soundcard that supports it.
*/
#include <CoreServices/CoreServices.h>
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/AudioToolbox.h>
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <inttypes.h>
#include <sys/types.h>
#include <unistd.h>
#include "config.h"
#include "core/mp_msg.h"
#include "ao.h"
#include "audio_out_internal.h"
#include "audio/format.h"
#include "osdep/timer.h"
#include "libavutil/fifo.h"
#include "core/subopt-helper.h"
static const ao_info_t info =
{
"Darwin/Mac OS X native audio output",
"coreaudio",
"Timothy J. Wood & Dan Christiansen & Chris Roccati",
""
};
LIBAO_EXTERN(coreaudio)
/* Prefix for all mp_msg() calls */
#define ao_msg(a, b, c...) mp_msg(a, b, "AO: [coreaudio] " c)
#if MAC_OS_X_VERSION_MAX_ALLOWED <= 1040
/* AudioDeviceIOProcID does not exist in Mac OS X 10.4. We can emulate
* this by using AudioDeviceAddIOProc() and AudioDeviceRemoveIOProc(). */
#define AudioDeviceIOProcID AudioDeviceIOProc
#define AudioDeviceDestroyIOProcID AudioDeviceRemoveIOProc
static OSStatus AudioDeviceCreateIOProcID(AudioDeviceID dev,
AudioDeviceIOProc proc,
void *data,
AudioDeviceIOProcID *procid)
{
*procid = proc;
return AudioDeviceAddIOProc(dev, proc, data);
}
#endif
typedef struct ao_coreaudio_s
{
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;
/* Ring-buffer */
AVFifoBuffer *buffer;
unsigned int buffer_len; ///< must always be num_chunks * chunk_size
unsigned int num_chunks;
unsigned int chunk_size;
} ao_coreaudio_t;
static ao_coreaudio_t *ao = NULL;
/**
* \brief add data to ringbuffer
*/
static int write_buffer(unsigned char* data, int len){
int free = ao->buffer_len - av_fifo_size(ao->buffer);
if (len > free) len = free;
return av_fifo_generic_write(ao->buffer, data, len, NULL);
}
/**
* \brief remove data from ringbuffer
*/
static int read_buffer(unsigned char* data,int len){
int buffered = av_fifo_size(ao->buffer);
if (len > buffered) len = buffered;
if (data)
av_fifo_generic_read(ao->buffer, data, len, NULL);
else
av_fifo_drain(ao->buffer, len);
return len;
}
static OSStatus theRenderProc(void *inRefCon,
AudioUnitRenderActionFlags *inActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber, UInt32 inNumFrames,
AudioBufferList *ioData)
{
int amt=av_fifo_size(ao->buffer);
int req=(inNumFrames)*ao->packetSize;
if(amt>req)
amt=req;
if(amt)
read_buffer((unsigned char *)ioData->mBuffers[0].mData, amt);
else audio_pause();
ioData->mBuffers[0].mDataByteSize = amt;
return noErr;
}
static int control(int cmd,void *arg){
ao_control_vol_t *control_vol;
OSStatus err;
Float32 vol;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
control_vol = (ao_control_vol_t*)arg;
if (ao->b_digital) {
// Digital output has no volume adjust.
int vol = ao->b_muted ? 0 : 100;
*control_vol = (ao_control_vol_t) {
.left = vol, .right = vol,
};
return CONTROL_TRUE;
}
err = AudioUnitGetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, &vol);
if(err==0) {
// printf("GET VOL=%f\n", vol);
control_vol->left=control_vol->right=vol*100.0/4.0;
return CONTROL_TRUE;
}
else {
ao_msg(MSGT_AO, MSGL_WARN, "could not get HAL output volume: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
case AOCONTROL_SET_VOLUME:
control_vol = (ao_control_vol_t*)arg;
if (ao->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)
ao->b_muted = 1;
else
ao->b_muted = 0;
return CONTROL_TRUE;
}
vol=(control_vol->left+control_vol->right)*4.0/200.0;
err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0);
if(err==0) {
// printf("SET VOL=%f\n", vol);
return CONTROL_TRUE;
}
else {
ao_msg(MSGT_AO, MSGL_WARN, "could not set HAL output volume: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
/* Everything is currently unimplemented */
default:
return CONTROL_FALSE;
}
}
static void print_format(int lev, const char* str, const AudioStreamBasicDescription *f){
uint32_t flags=(uint32_t) f->mFormatFlags;
ao_msg(MSGT_AO,lev, "%s %7.1fHz %"PRIu32"bit [%c%c%c%c][%"PRIu32"][%"PRIu32"][%"PRIu32"][%"PRIu32"][%"PRIu32"] %s %s %s%s%s%s\n",
str, f->mSampleRate, f->mBitsPerChannel,
(int)(f->mFormatID & 0xff000000) >> 24,
(int)(f->mFormatID & 0x00ff0000) >> 16,
(int)(f->mFormatID & 0x0000ff00) >> 8,
(int)(f->mFormatID & 0x000000ff) >> 0,
f->mFormatFlags, f->mBytesPerPacket,
f->mFramesPerPacket, f->mBytesPerFrame,
f->mChannelsPerFrame,
(flags&kAudioFormatFlagIsFloat) ? "float" : "int",
(flags&kAudioFormatFlagIsBigEndian) ? "BE" : "LE",
(flags&kAudioFormatFlagIsSignedInteger) ? "S" : "U",
(flags&kAudioFormatFlagIsPacked) ? " packed" : "",
(flags&kAudioFormatFlagIsAlignedHigh) ? " aligned" : "",
(flags&kAudioFormatFlagIsNonInterleaved) ? " ni" : "" );
}
static OSStatus GetAudioProperty(AudioObjectID id,
AudioObjectPropertySelector selector,
UInt32 outSize, void *outData)
{
AudioObjectPropertyAddress property_address;
property_address.mSelector = selector;
property_address.mScope = kAudioObjectPropertyScopeGlobal;
property_address.mElement = kAudioObjectPropertyElementMaster;
return AudioObjectGetPropertyData(id, &property_address, 0, NULL, &outSize, outData);
}
static UInt32 GetAudioPropertyArray(AudioObjectID id,
AudioObjectPropertySelector selector,
AudioObjectPropertyScope scope,
void **outData)
{
OSStatus err;
AudioObjectPropertyAddress property_address;
UInt32 i_param_size;
property_address.mSelector = selector;
property_address.mScope = scope;
property_address.mElement = kAudioObjectPropertyElementMaster;
err = AudioObjectGetPropertyDataSize(id, &property_address, 0, NULL, &i_param_size);
if (err != noErr)
return 0;
*outData = malloc(i_param_size);
err = AudioObjectGetPropertyData(id, &property_address, 0, NULL, &i_param_size, *outData);
if (err != noErr) {
free(*outData);
return 0;
}
return i_param_size;
}
static UInt32 GetGlobalAudioPropertyArray(AudioObjectID id,
AudioObjectPropertySelector selector,
void **outData)
{
return GetAudioPropertyArray(id, selector, kAudioObjectPropertyScopeGlobal, outData);
}
static OSStatus GetAudioPropertyString(AudioObjectID id,
AudioObjectPropertySelector selector,
char **outData)
{
OSStatus err;
AudioObjectPropertyAddress property_address;
UInt32 i_param_size;
CFStringRef string;
CFIndex string_length;
property_address.mSelector = selector;
property_address.mScope = kAudioObjectPropertyScopeGlobal;
property_address.mElement = kAudioObjectPropertyElementMaster;
i_param_size = sizeof(CFStringRef);
err = AudioObjectGetPropertyData(id, &property_address, 0, NULL, &i_param_size, &string);
if (err != noErr)
return err;
string_length = CFStringGetMaximumSizeForEncoding(CFStringGetLength(string),
kCFStringEncodingASCII);
*outData = malloc(string_length + 1);
CFStringGetCString(string, *outData, string_length + 1, kCFStringEncodingASCII);
CFRelease(string);
return err;
}
static OSStatus SetAudioProperty(AudioObjectID id,
AudioObjectPropertySelector selector,
UInt32 inDataSize, void *inData)
{
AudioObjectPropertyAddress property_address;
property_address.mSelector = selector;
property_address.mScope = kAudioObjectPropertyScopeGlobal;
property_address.mElement = kAudioObjectPropertyElementMaster;
return AudioObjectSetPropertyData(id, &property_address, 0, NULL, inDataSize, inData);
}
static Boolean IsAudioPropertySettable(AudioObjectID id,
AudioObjectPropertySelector selector,
Boolean *outData)
{
AudioObjectPropertyAddress property_address;
property_address.mSelector = selector;
property_address.mScope = kAudioObjectPropertyScopeGlobal;
property_address.mElement = kAudioObjectPropertyElementMaster;
return AudioObjectIsPropertySettable(id, &property_address, outData);
}
static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id );
static int AudioStreamSupportsDigital( AudioStreamID i_stream_id );
static int OpenSPDIF(void);
static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format );
static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
const AudioTimeStamp * inNow,
const void * inInputData,
const AudioTimeStamp * inInputTime,
AudioBufferList * outOutputData,
const AudioTimeStamp * inOutputTime,
void * threadGlobals );
static OSStatus StreamListener( AudioObjectID inObjectID,
UInt32 inNumberAddresses,
const AudioObjectPropertyAddress inAddresses[],
void *inClientData );
static OSStatus DeviceListener( AudioObjectID inObjectID,
UInt32 inNumberAddresses,
const AudioObjectPropertyAddress inAddresses[],
void *inClientData );
static void print_help(void)
{
OSStatus err;
UInt32 i_param_size;
int num_devices;
AudioDeviceID *devids;
char *device_name;
mp_msg(MSGT_AO, 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");
i_param_size = GetGlobalAudioPropertyArray(kAudioObjectSystemObject, kAudioHardwarePropertyDevices, (void **)&devids);
if (!i_param_size) {
mp_msg(MSGT_AO, MSGL_FATAL, "Failed to get list of output devices.\n");
return;
}
num_devices = i_param_size / sizeof(AudioDeviceID);
for (int i = 0; i < num_devices; ++i) {
err = GetAudioPropertyString(devids[i], kAudioObjectPropertyName, &device_name);
if (err == noErr) {
mp_msg(MSGT_AO, MSGL_FATAL, "%s (id: %"PRIu32")\n", device_name, devids[i]);
free(device_name);
} else
mp_msg(MSGT_AO, MSGL_FATAL, "Unknown (id: %"PRIu32")\n", devids[i]);
}
mp_msg(MSGT_AO, MSGL_FATAL, "\n");
free(devids);
}
static int init(int rate,int channels,int format,int flags)
{
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, display_help = 0;
const opt_t subopts[] = {
{"device_id", OPT_ARG_INT, &device_id, NULL},
{"help", OPT_ARG_BOOL, &display_help, NULL},
{NULL}
};
// set defaults
device_id = 0;
if (subopt_parse(ao_subdevice, subopts) != 0 || display_help) {
print_help();
if (!display_help)
return 0;
}
ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);
ao = calloc(1, sizeof(ao_coreaudio_t));
ao->i_selected_dev = 0;
ao->b_supports_digital = 0;
ao->b_digital = 0;
ao->b_muted = 0;
ao->b_stream_format_changed = 0;
ao->i_hog_pid = -1;
ao->i_stream_id = 0;
ao->i_stream_index = -1;
ao->b_revert = 0;
ao->b_changed_mixing = 0;
global_ao->per_application_mixer = true;
global_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)
{
ao_msg(MSGT_AO, 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)
{
ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
goto err_out;
}
ao_msg(MSGT_AO,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(format)) {
if (AudioDeviceSupportsDigital(devid_def))
{
ao->b_supports_digital = 1;
}
ao_msg(MSGT_AO, MSGL_V,
"probe default audio output device about support for digital s/pdif output: %d\n",
ao->b_supports_digital );
}
free(psz_name);
// Save selected device id
ao->i_selected_dev = devid_def;
// Build Description for the input format
inDesc.mSampleRate=rate;
inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
inDesc.mChannelsPerFrame=channels;
inDesc.mBitsPerChannel=af_fmt2bits(format);
if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
// float
inDesc.mFormatFlags = kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked;
}
else if((format&AF_FORMAT_SIGN_MASK)==AF_FORMAT_SI) {
// signed int
inDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
}
else {
// unsigned int
inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
}
if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
inDesc.mFramesPerPacket = 1;
ao->packetSize = inDesc.mBytesPerPacket = inDesc.mBytesPerFrame = inDesc.mFramesPerPacket*channels*(inDesc.mBitsPerChannel/8);
print_format(MSGL_V, "source:",&inDesc);
if (ao->b_supports_digital)
{
b_alive = 1;
err = GetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertyDeviceIsAlive,
sizeof(UInt32), &b_alive);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is alive: [%4.4s]\n", (char *)&err);
if (!b_alive)
ao_msg(MSGT_AO, MSGL_WARN, "device is not alive\n" );
/* S/PDIF output need device in HogMode. */
err = GetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(pid_t), &ao->i_hog_pid);
if (err != noErr)
{
/* This is not a fatal error. Some drivers simply don't support this property. */
ao_msg(MSGT_AO, MSGL_WARN, "could not check whether device is hogged: [%4.4s]\n",
(char *)&err);
ao->i_hog_pid = -1;
}
if (ao->i_hog_pid != -1 && ao->i_hog_pid != getpid())
{
ao_msg(MSGT_AO, MSGL_WARN, "Selected audio device is exclusively in use by another program.\n" );
goto err_out;
}
ao->stream_format = inDesc;
return OpenSPDIF();
}
/* 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) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
goto err_out;
}
err = AudioComponentInstanceNew(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
goto err_out;
}
// Initialize AudioUnit
err = AudioUnitInitialize(ao->theOutputUnit);
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
goto err_out1;
}
size = sizeof(AudioStreamBasicDescription);
err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &inDesc, size);
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the input format: [%4.4s]\n", (char *)&err);
goto err_out2;
}
size = sizeof(UInt32);
err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
if (err)
{
ao_msg(MSGT_AO,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(ao->theOutputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &ao->i_selected_dev, sizeof(ao->i_selected_dev));
ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
ao_data.samplerate = inDesc.mSampleRate;
ao_data.channels = inDesc.mChannelsPerFrame;
ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
ao_data.outburst = ao->chunk_size;
ao_data.buffersize = ao_data.bps;
ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
ao->buffer_len = ao->num_chunks * ao->chunk_size;
ao->buffer = av_fifo_alloc(ao->buffer_len);
ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
renderCallback.inputProc = theRenderProc;
renderCallback.inputProcRefCon = 0;
err = AudioUnitSetProperty(ao->theOutputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &renderCallback, sizeof(AURenderCallbackStruct));
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to set the render callback: [%4.4s]\n", (char *)&err);
goto err_out2;
}
reset();
return CONTROL_OK;
err_out2:
AudioUnitUninitialize(ao->theOutputUnit);
err_out1:
AudioComponentInstanceDispose(ao->theOutputUnit);
err_out:
av_fifo_free(ao->buffer);
free(ao);
ao = NULL;
return CONTROL_FALSE;
}
/*****************************************************************************
* Setup a encoded digital stream (SPDIF)
*****************************************************************************/
static int OpenSPDIF(void)
{
OSStatus err = noErr;
UInt32 i_param_size, b_mix = 0;
Boolean b_writeable = 0;
AudioStreamID *p_streams = NULL;
int i, i_streams = 0;
AudioObjectPropertyAddress property_address;
/* Start doing the SPDIF setup process. */
ao->b_digital = 1;
/* Hog the device. */
ao->i_hog_pid = getpid() ;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(ao->i_hog_pid), &ao->i_hog_pid);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
ao->i_hog_pid = -1;
goto err_out;
}
property_address.mSelector = kAudioDevicePropertySupportsMixing;
property_address.mScope = kAudioObjectPropertyScopeGlobal;
property_address.mElement = kAudioObjectPropertyElementMaster;
/* Set mixable to false if we are allowed to. */
if (AudioObjectHasProperty(ao->i_selected_dev, &property_address)) {
/* Set mixable to false if we are allowed to. */
err = IsAudioPropertySettable(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
&b_writeable);
err = GetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
if (err == noErr && b_writeable)
{
b_mix = 0;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
ao->b_changed_mixing = 1;
}
if (err != noErr)
{
ao_msg(MSGT_AO, 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(ao->i_selected_dev,
kAudioDevicePropertyStreams,
kAudioDevicePropertyScopeOutput,
(void **)&p_streams);
if (!i_param_size) {
ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n");
goto err_out;
}
i_streams = i_param_size / sizeof(AudioStreamID);
ao_msg(MSGT_AO, MSGL_V, "current device stream number: %d\n", i_streams);
for (i = 0; i < i_streams && ao->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) {
ao_msg(MSGT_AO, 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;
ao->i_stream_id = p_streams[i];
ao->i_stream_index = i;
if (ao->b_revert == 0)
{
/* Retrieve the original format of this stream first if not done so already. */
err = GetAudioProperty(ao->i_stream_id,
kAudioStreamPropertyPhysicalFormat,
sizeof(ao->sfmt_revert), &ao->sfmt_revert);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN,
"Could not retrieve the original stream format: [%4.4s]\n",
(char *)&err);
free(p_format_list);
continue;
}
ao->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 == ao->stream_format.mSampleRate)
{
i_requested_rate_format = j;
break;
}
if (p_format_list[j].mFormat.mSampleRate == ao->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. */
ao->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. */
ao->stream_format = p_format_list[i_current_rate_format].mFormat;
else ao->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 (ao->i_stream_index < 0)
{
ao_msg(MSGT_AO, MSGL_WARN,
"Cannot find any digital output stream format when OpenSPDIF().\n");
goto err_out;
}
print_format(MSGL_V, "original stream format:", &ao->sfmt_revert);
if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
goto err_out;
property_address.mSelector = kAudioDevicePropertyDeviceHasChanged;
property_address.mScope = kAudioObjectPropertyScopeGlobal;
property_address.mElement = kAudioObjectPropertyElementMaster;
err = AudioObjectAddPropertyListener(ao->i_selected_dev,
&property_address,
DeviceListener,
NULL);
if (err != noErr)
ao_msg(MSGT_AO, 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 (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
#else
/* tell mplayer that we need a byteswap on AC3 streams, */
if (ao->stream_format.mFormatID & kAudioFormat60958AC3)
ao_data.format = AF_FORMAT_AC3_LE;
if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
ao_msg(MSGT_AO, MSGL_WARN,
"Output stream has non-native byte order, digital output may fail.\n");
/* For ac3/dts, just use packet size 6144 bytes as chunk size. */
ao->chunk_size = ao->stream_format.mBytesPerPacket;
ao_data.samplerate = ao->stream_format.mSampleRate;
ao_data.channels = ao->stream_format.mChannelsPerFrame;
ao_data.bps = ao_data.samplerate * (ao->stream_format.mBytesPerPacket/ao->stream_format.mFramesPerPacket);
ao_data.outburst = ao->chunk_size;
ao_data.buffersize = ao_data.bps;
ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
ao->buffer_len = ao->num_chunks * ao->chunk_size;
ao->buffer = av_fifo_alloc(ao->buffer_len);
ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
/* Create IOProc callback. */
err = AudioDeviceCreateIOProcID(ao->i_selected_dev,
(AudioDeviceIOProc)RenderCallbackSPDIF,
(void *)ao,
&ao->renderCallback);
if (err != noErr || ao->renderCallback == NULL)
{
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err);
goto err_out1;
}
reset();
return CONTROL_TRUE;
err_out1:
if (ao->b_revert)
AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
err_out:
if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
{
int b_mix = 1;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(int), &b_mix);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n",
(char *)&err);
}
if (ao->i_hog_pid == getpid())
{
ao->i_hog_pid = -1;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(ao->i_hog_pid), &ao->i_hog_pid);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n",
(char *)&err);
}
av_fifo_free(ao->buffer);
free(ao);
ao = NULL;
return CONTROL_FALSE;
}
/*****************************************************************************
* AudioDeviceSupportsDigital: Check i_dev_id for digital stream support.
*****************************************************************************/
static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id )
{
UInt32 i_param_size = 0;
AudioStreamID *p_streams = NULL;
int i = 0, i_streams = 0;
int b_return = CONTROL_FALSE;
/* Retrieve all the output streams. */
i_param_size = GetAudioPropertyArray(i_dev_id,
kAudioDevicePropertyStreams,
kAudioDevicePropertyScopeOutput,
(void **)&p_streams);
if (!i_param_size) {
ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams.\n");
return CONTROL_FALSE;
}
i_streams = i_param_size / sizeof(AudioStreamID);
for (i = 0; i < i_streams; ++i)
{
if (AudioStreamSupportsDigital(p_streams[i]))
b_return = CONTROL_OK;
}
free(p_streams);
return b_return;
}
/*****************************************************************************
* AudioStreamSupportsDigital: Check i_stream_id for digital stream support.
*****************************************************************************/
static int AudioStreamSupportsDigital( AudioStreamID i_stream_id )
{
UInt32 i_param_size;
AudioStreamRangedDescription *p_format_list = NULL;
int i, i_formats, b_return = CONTROL_FALSE;
/* Retrieve all the stream formats supported by each output stream. */
i_param_size = GetGlobalAudioPropertyArray(i_stream_id,
kAudioStreamPropertyAvailablePhysicalFormats,
(void **)&p_format_list);
if (!i_param_size) {
ao_msg(MSGT_AO, MSGL_WARN, "Could not get number of stream formats.\n");
return CONTROL_FALSE;
}
i_formats = i_param_size / sizeof(AudioStreamRangedDescription);
for (i = 0; i < i_formats; ++i)
{
print_format(MSGL_V, "supported format:", &(p_format_list[i].mFormat));
if (p_format_list[i].mFormat.mFormatID == 'IAC3' ||
p_format_list[i].mFormat.mFormatID == 'iac3' ||
p_format_list[i].mFormat.mFormatID == kAudioFormat60958AC3 ||
p_format_list[i].mFormat.mFormatID == kAudioFormatAC3)
b_return = CONTROL_OK;
}
free(p_format_list);
return b_return;
}
/*****************************************************************************
* AudioStreamChangeFormat: Change i_stream_id to change_format
*****************************************************************************/
static int AudioStreamChangeFormat( AudioStreamID i_stream_id, AudioStreamBasicDescription change_format )
{
OSStatus err = noErr;
int i;
AudioObjectPropertyAddress property_address;
static volatile int stream_format_changed;
stream_format_changed = 0;
print_format(MSGL_V, "setting stream format:", &change_format);
/* Install the callback. */
property_address.mSelector = kAudioStreamPropertyPhysicalFormat;
property_address.mScope = kAudioObjectPropertyScopeGlobal;
property_address.mElement = kAudioObjectPropertyElementMaster;
err = AudioObjectAddPropertyListener(i_stream_id,
&property_address,
StreamListener,
(void *)&stream_format_changed);
if (err != noErr)
{
ao_msg(MSGT_AO, 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)
{
ao_msg(MSGT_AO, 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)
usec_sleep(10000);
if (stream_format_changed)
stream_format_changed = 0;
else
ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" );
err = GetAudioProperty(i_stream_id,
kAudioStreamPropertyPhysicalFormat,
sizeof(AudioStreamBasicDescription), &actual_format);
print_format(MSGL_V, "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,
&property_address,
StreamListener,
(void *)&stream_format_changed);
if (err != noErr)
{
ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err);
return CONTROL_FALSE;
}
return CONTROL_TRUE;
}
/*****************************************************************************
* RenderCallbackSPDIF: callback for SPDIF audio output
*****************************************************************************/
static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
const AudioTimeStamp * inNow,
const void * inInputData,
const AudioTimeStamp * inInputTime,
AudioBufferList * outOutputData,
const AudioTimeStamp * inOutputTime,
void * threadGlobals )
{
int amt = av_fifo_size(ao->buffer);
int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize;
if (amt > req)
amt = req;
if (amt)
read_buffer(ao->b_muted ? NULL : (unsigned char *)outOutputData->mBuffers[ao->i_stream_index].mData, amt);
return noErr;
}
static int play(void* output_samples,int num_bytes,int flags)
{
int wrote, b_digital;
SInt32 exit_reason;
// Check whether we need to reset the digital output stream.
if (ao->b_digital && ao->b_stream_format_changed)
{
ao->b_stream_format_changed = 0;
b_digital = AudioStreamSupportsDigital(ao->i_stream_id);
if (b_digital)
{
/* Current stream supports digital format output, let's set it. */
ao_msg(MSGT_AO, MSGL_V,
"Detected current stream supports digital, try to restore digital output...\n");
if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
{
ao_msg(MSGT_AO, MSGL_WARN, "Restoring digital output failed.\n");
}
else
{
ao_msg(MSGT_AO, MSGL_WARN, "Restoring digital output succeeded.\n");
reset();
}
}
else
ao_msg(MSGT_AO, MSGL_V, "Detected current stream does not support digital.\n");
}
wrote=write_buffer(output_samples, num_bytes);
audio_resume();
do {
exit_reason = CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0, true);
} while (exit_reason == kCFRunLoopRunHandledSource);
return wrote;
}
/* set variables and buffer to initial state */
static void reset(void)
{
audio_pause();
av_fifo_reset(ao->buffer);
}
/* return available space */
static int get_space(void)
{
return ao->buffer_len - av_fifo_size(ao->buffer);
}
/* return delay until audio is played */
static float get_delay(void)
{
// inaccurate, should also contain the data buffered e.g. by the OS
return (float)av_fifo_size(ao->buffer)/(float)ao_data.bps;
}
/* unload plugin and deregister from coreaudio */
static void uninit(int immed)
{
OSStatus err = noErr;
if (!immed) {
long long timeleft=(1000000LL*av_fifo_size(ao->buffer))/ao_data.bps;
ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", av_fifo_size(ao->buffer), ao_data.bps, (int)timeleft);
usec_sleep((int)timeleft);
}
if (!ao->b_digital) {
AudioOutputUnitStop(ao->theOutputUnit);
AudioUnitUninitialize(ao->theOutputUnit);
AudioComponentInstanceDispose(ao->theOutputUnit);
}
else {
/* Stop device. */
err = AudioDeviceStop(ao->i_selected_dev, ao->renderCallback);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
/* Remove IOProc callback. */
err = AudioDeviceDestroyIOProcID(ao->i_selected_dev, ao->renderCallback);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceRemoveIOProc failed: [%4.4s]\n", (char *)&err);
if (ao->b_revert)
AudioStreamChangeFormat(ao->i_stream_id, ao->sfmt_revert);
if (ao->b_changed_mixing && ao->sfmt_revert.mFormatID != kAudioFormat60958AC3)
{
UInt32 b_mix;
Boolean b_writeable = 0;
/* Revert mixable to true if we are allowed to. */
err = IsAudioPropertySettable(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
&b_writeable);
err = GetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
if (err == noErr && b_writeable)
{
b_mix = 1;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertySupportsMixing,
sizeof(UInt32), &b_mix);
}
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "failed to set mixmode: [%4.4s]\n", (char *)&err);
}
if (ao->i_hog_pid == getpid())
{
ao->i_hog_pid = -1;
err = SetAudioProperty(ao->i_selected_dev,
kAudioDevicePropertyHogMode,
sizeof(ao->i_hog_pid), &ao->i_hog_pid);
if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err);
}
}
av_fifo_free(ao->buffer);
free(ao);
ao = NULL;
}
/* stop playing, keep buffers (for pause) */
static void audio_pause(void)
{
OSErr err=noErr;
/* Stop callback. */
if (!ao->b_digital)
{
err=AudioOutputUnitStop(ao->theOutputUnit);
if (err != noErr)
ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStop returned [%4.4s]\n", (char *)&err);
}
else
{
err = AudioDeviceStop(ao->i_selected_dev, ao->renderCallback);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);
}
ao->paused = 1;
}
/* resume playing, after audio_pause() */
static void audio_resume(void)
{
OSErr err=noErr;
if (!ao->paused)
return;
/* Start callback. */
if (!ao->b_digital)
{
err = AudioOutputUnitStart(ao->theOutputUnit);
if (err != noErr)
ao_msg(MSGT_AO,MSGL_WARN, "AudioOutputUnitStart returned [%4.4s]\n", (char *)&err);
}
else
{
err = AudioDeviceStart(ao->i_selected_dev, ao->renderCallback);
if (err != noErr)
ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err);
}
ao->paused = 0;
}
/*****************************************************************************
* StreamListener
*****************************************************************************/
static OSStatus StreamListener( AudioObjectID inObjectID,
UInt32 inNumberAddresses,
const AudioObjectPropertyAddress inAddresses[],
void *inClientData )
{
for (int i=0; i < inNumberAddresses; ++i)
{
if (inAddresses[i].mSelector == kAudioStreamPropertyPhysicalFormat) {
ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioStreamPropertyPhysicalFormat changed.\n");
if (inClientData)
*(volatile int *)inClientData = 1;
break;
}
}
return noErr;
}
static OSStatus DeviceListener( AudioObjectID inObjectID,
UInt32 inNumberAddresses,
const AudioObjectPropertyAddress inAddresses[],
void *inClientData )
{
for (int i=0; i < inNumberAddresses; ++i)
{
if (inAddresses[i].mSelector == kAudioDevicePropertyDeviceHasChanged) {
ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n");
ao->b_stream_format_changed = 1;
break;
}
}
return noErr;
}
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