view libao2/ao_macosx.c @ 24787:02535b3216c5

Avoid text deformation and subtitles moving outside the screen in pan-and-scan mode. For this, crop amounts are passed from vo_gl as negative margins sizes. They are used to calculate aspect ratio. They are ignored when calculating subtitle positions, so subtitles will stay on screen most of the time. Based on a patch by Jindrich Makovicka [makovick gmail com].

/*
 *
 *  ao_macosx.c
 *
 *      Original Copyright (C) Timothy J. Wood - Aug 2000
 *
 *  This file is part of libao, a cross-platform library.  See
 *  README for a history of this source code.
 *
 *  libao 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, or (at your option)
 *  any later version.
 *
 *  libao is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License along
 *  with libao; 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). 
 */

/* Change log:
 * 
 * 14/5-2003: Ported to MPlayer libao2 by Dan Christiansen
 *
 *            AC-3 and MPEG audio passthrough is possible, but I don't have
 *            access to a sound card 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 <pthread.h>
#include <sys/types.h>
#include <unistd.h>

#include "config.h"
#include "mp_msg.h"

#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "osdep/timer.h"

static ao_info_t info =
  {
    "Darwin/Mac OS X native audio output",
    "macosx",
    "Timothy J. Wood & Dan Christiansen & Chris Roccati",
    ""
  };

LIBAO_EXTERN(macosx)

/* Prefix for all mp_msg() calls */
#define ao_msg(a, b, c...) mp_msg(a, b, "AO: [macosx] " c)

/* This is large, but best (maybe it should be even larger).
 * CoreAudio supposedly has an internal latency in the order of 2ms */
#define NUM_BUFS 32

typedef struct ao_macosx_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? */

  /* 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 */
  /* does not need explicit synchronization, but needs to allocate
   * (num_chunks + 1) * chunk_size memory to store num_chunks * chunk_size
   * data */
  unsigned char *buffer;
  unsigned int buffer_len; ///< must always be (num_chunks + 1) * chunk_size
  unsigned int num_chunks;
  unsigned int chunk_size;
  
  unsigned int buf_read_pos;
  unsigned int buf_write_pos;
} ao_macosx_t;

static ao_macosx_t *ao = NULL;

/**
 * \brief return number of free bytes in the buffer
 *    may only be called by mplayer's thread
 * \return minimum number of free bytes in buffer, value may change between
 *    two immediately following calls, and the real number of free bytes
 *    might actually be larger!
 */
static int buf_free(void) {
  int free = ao->buf_read_pos - ao->buf_write_pos - ao->chunk_size;
  if (free < 0) free += ao->buffer_len;
  return free;
}

/**
 * \brief return number of buffered bytes
 *    may only be called by playback thread
 * \return minimum number of buffered bytes, value may change between
 *    two immediately following calls, and the real number of buffered bytes
 *    might actually be larger!
 */
static int buf_used(void) {
  int used = ao->buf_write_pos - ao->buf_read_pos;
  if (used < 0) used += ao->buffer_len;
  return used;
}

/**
 * \brief add data to ringbuffer
 */
static int write_buffer(unsigned char* data, int len){
  int first_len = ao->buffer_len - ao->buf_write_pos;
  int free = buf_free();
  if (len > free) len = free;
  if (first_len > len) first_len = len;
  // till end of buffer
  memcpy (&ao->buffer[ao->buf_write_pos], data, first_len);
  if (len > first_len) { // we have to wrap around
    // remaining part from beginning of buffer
    memcpy (ao->buffer, &data[first_len], len - first_len);
  }
  ao->buf_write_pos = (ao->buf_write_pos + len) % ao->buffer_len;
  return len;
}

/**
 * \brief remove data from ringbuffer
 */
static int read_buffer(unsigned char* data,int len){
  int first_len = ao->buffer_len - ao->buf_read_pos;
  int buffered = buf_used();
  if (len > buffered) len = buffered;
  if (first_len > len) first_len = len;
  // till end of buffer
  memcpy (data, &ao->buffer[ao->buf_read_pos], first_len);
  if (len > first_len) { // we have to wrap around
    // remaining part from beginning of buffer
    memcpy (&data[first_len], ao->buffer, len - first_len);
  }
  ao->buf_read_pos = (ao->buf_read_pos + len) % ao->buffer_len;
  return len;
}

OSStatus theRenderProc(void *inRefCon, AudioUnitRenderActionFlags *inActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumFrames, AudioBufferList *ioData)
{
int amt=buf_used();
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.
			return CONTROL_FALSE;
		}
		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:
		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.
			return CONTROL_TRUE;
		control_vol = (ao_control_vol_t*)arg;
		
		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 %lubit [%c%c%c%c][%lu][%lu][%lu][%lu][%lu] %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 int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id );
static int AudioStreamSupportsDigital( AudioStreamID i_stream_id );
static int OpenSPDIF();
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( AudioStreamID inStream,
                                UInt32 inChannel,
                                AudioDevicePropertyID inPropertyID,
                                void * inClientData );
static OSStatus DeviceListener( AudioDeviceID inDevice,
                                UInt32 inChannel,
                                Boolean isInput,
                                AudioDevicePropertyID inPropertyID,
                                void* inClientData );

static int init(int rate,int channels,int format,int flags)
{
AudioStreamBasicDescription inDesc;
ComponentDescription desc; 
Component comp; 
AURenderCallbackStruct renderCallback;
OSStatus err;
UInt32 size, maxFrames, i_param_size;
char *psz_name;
int aoIsCreated = ao != NULL;
AudioDeviceID devid_def = 0;
int b_alive;

    ao_msg(MSGT_AO,MSGL_V, "init([%dHz][%dch][%s][%d])\n", rate, channels, af_fmt2str_short(format), flags);

    if (!aoIsCreated) { ao = malloc(sizeof(ao_macosx_t)); ao->buffer = NULL;}

    ao->i_selected_dev = 0;
    ao->b_supports_digital = 0;
    ao->b_digital = 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;

    /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
    if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3)
    {
        /* Find the ID of the default Device. */
        i_param_size = sizeof(AudioDeviceID);
        err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
                                       &i_param_size, &devid_def);
        if (err != noErr)
        {
            ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device: [%4.4s]\n", (char *)&err);
            return CONTROL_FALSE;
        }

        /* Retrieve the length of the device name. */
        i_param_size = 0;
        err = AudioDeviceGetPropertyInfo(devid_def, 0, 0,
                                         kAudioDevicePropertyDeviceName,
                                         &i_param_size, NULL);
        if (err != noErr)
        {
            ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name length: [%4.4s]\n", (char *)&err);
            return CONTROL_FALSE;
        }

        /* Retrieve the name of the device. */
        psz_name = (char *)malloc(i_param_size);
        err = AudioDeviceGetProperty(devid_def, 0, 0,
                                     kAudioDevicePropertyDeviceName,
                                     &i_param_size, psz_name);
        if (err != noErr)
        {
            ao_msg(MSGT_AO, MSGL_WARN, "could not get default audio device name: [%4.4s]\n", (char *)&err);
            return CONTROL_FALSE;
        }

        ao_msg(MSGT_AO,MSGL_V, "got default audio output device ID: %#lx Name: %s\n", devid_def, psz_name );

        if (AudioDeviceSupportsDigital(devid_def))
        {
            ao->b_supports_digital = 1;
            ao->i_selected_dev = devid_def;
        }
        ao_msg(MSGT_AO,MSGL_V, "probe default audio output device whether support digital s/pdif output:%d\n", ao->b_supports_digital );

        free( psz_name);
    }

	// Build Description for the input format
	inDesc.mSampleRate=rate;
	inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
	inDesc.mChannelsPerFrame=channels;
	switch(format&AF_FORMAT_BITS_MASK){
	case AF_FORMAT_8BIT:
		inDesc.mBitsPerChannel=8;
		break;
	case AF_FORMAT_16BIT:
		inDesc.mBitsPerChannel=16;
		break;
	case AF_FORMAT_24BIT:
		inDesc.mBitsPerChannel=24;
		break;
	case AF_FORMAT_32BIT:
		inDesc.mBitsPerChannel=32;
		break;
	default:
		ao_msg(MSGT_AO, MSGL_WARN, "Unsupported format (0x%08x)\n", format);
		return CONTROL_FALSE;
		break;
	}

    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_SPECIAL_MASK) == AF_FORMAT_AC3) {
        // Currently ac3 input (comes from hwac3) is always in native byte-order.
#ifdef WORDS_BIGENDIAN
        inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
#endif
    }
    else 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;
        i_param_size = sizeof(b_alive);
        err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
                                     kAudioDevicePropertyDeviceIsAlive,
                                     &i_param_size, &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. */
        i_param_size = sizeof(ao->i_hog_pid);
        err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
                                     kAudioDevicePropertyHogMode,
                                     &i_param_size, &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" );
            return CONTROL_FALSE;
        }
        ao->stream_format = inDesc;
        return OpenSPDIF();
    }

	/* original analog output code */
	if (!aoIsCreated) {
	desc.componentType = kAudioUnitType_Output;
	desc.componentSubType = kAudioUnitSubType_DefaultOutput;
	desc.componentManufacturer = kAudioUnitManufacturer_Apple;
	desc.componentFlags = 0;
	desc.componentFlagsMask = 0;
				
	comp = FindNextComponent(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");
		return CONTROL_FALSE;
	}
		
	err = OpenAComponent(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);
		return CONTROL_FALSE;
	}

	// 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);
		return CONTROL_FALSE;
	}
	}

	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);
		return CONTROL_FALSE;
	}

	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);
		return CONTROL_FALSE;
	}

	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 + 1) * ao->chunk_size;
    ao->buffer = aoIsCreated ? realloc(ao->buffer,(ao->num_chunks + 1)*ao->chunk_size)
							: calloc(ao->num_chunks + 1, ao->chunk_size);
	
	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);
		return CONTROL_FALSE;
	}

	reset();
    
    return CONTROL_OK;
}

/*****************************************************************************
 * Setup a encoded digital stream (SPDIF)
 *****************************************************************************/
static int OpenSPDIF()
{
    OSStatus                err = noErr;
    UInt32                  i_param_size, b_mix = 0;
    Boolean                 b_writeable = 0;
    AudioStreamID           *p_streams = NULL;
    int                     i, i_streams = 0;

    /* Start doing the SPDIF setup process. */
    ao->b_digital = 1;

    /* Hog the device. */
    i_param_size = sizeof(ao->i_hog_pid);
    ao->i_hog_pid = getpid() ;

    err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
                                 kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);

    if (err != noErr)
    {
        ao_msg(MSGT_AO, MSGL_WARN, "failed to set hogmode: [%4.4s]\n", (char *)&err);
        return CONTROL_FALSE;
    }

    /* Set mixable to false if we are allowed to. */
    err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
                                     kAudioDevicePropertySupportsMixing,
                                     &i_param_size, &b_writeable);
    err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
                                 kAudioDevicePropertySupportsMixing,
                                 &i_param_size, &b_mix);
    if (err != noErr && b_writeable)
    {
        b_mix = 0;
        err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
                                     kAudioDevicePropertySupportsMixing,
                                     i_param_size, &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);
        return CONTROL_FALSE;
    }

    /* Get a list of all the streams on this device. */
    err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE,
                                     kAudioDevicePropertyStreams,
                                     &i_param_size, NULL);
    if (err != noErr)
    {
        ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
        return CONTROL_FALSE;
    }

    i_streams = i_param_size / sizeof(AudioStreamID);
    p_streams = (AudioStreamID *)malloc(i_param_size);
    if (p_streams == NULL)
    {
        ao_msg(MSGT_AO, MSGL_WARN, "out of memory\n" );
        return CONTROL_FALSE;
    }

    err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE,
                                 kAudioDevicePropertyStreams,
                                 &i_param_size, p_streams);
    if (err != noErr)
    {
        ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streams: [%4.4s]\n", (char *)&err);
        if (p_streams) free(p_streams);
        return CONTROL_FALSE;
    }

    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. */
        AudioStreamBasicDescription *p_format_list = NULL;
        int i_formats = 0, j = 0, b_digital = 0;

        /* Retrieve all the stream formats supported by each output stream. */
        err = AudioStreamGetPropertyInfo(p_streams[i], 0,
                                         kAudioStreamPropertyPhysicalFormats,
                                         &i_param_size, NULL);
        if (err != noErr)
        {
            ao_msg(MSGT_AO, MSGL_WARN, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
            continue;
        }

        i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
        p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
        if (p_format_list == NULL)
        {
            ao_msg(MSGT_AO, MSGL_WARN, "could not malloc the memory\n" );
            continue;
        }

        err = AudioStreamGetProperty(p_streams[i], 0,
                                     kAudioStreamPropertyPhysicalFormats,
                                     &i_param_size, p_format_list);
        if (err != noErr)
        {
            ao_msg(MSGT_AO, MSGL_WARN, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
            if (p_format_list) free(p_format_list);
            continue;
        }

        /* Check if one of the supported formats is a digital format. */
        for (j = 0; j < i_formats; ++j)
        {
            if (p_format_list[j].mFormatID == 'IAC3' ||
                  p_format_list[j].mFormatID == kAudioFormat60958AC3)
            {
                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. */
                i_param_size = sizeof(ao->sfmt_revert);
                err = AudioStreamGetProperty(ao->i_stream_id, 0,
                                             kAudioStreamPropertyPhysicalFormat,
                                             &i_param_size,
                                             &ao->sfmt_revert);
                if (err != noErr)
                {
                    ao_msg(MSGT_AO, MSGL_WARN, "could not retrieve the original streamformat: [%4.4s]\n", (char *)&err);
                    if (p_format_list) free(p_format_list);
                    continue;
                }
                ao->b_revert = 1;
            }

            for (j = 0; j < i_formats; ++j)
                if (p_format_list[j].mFormatID == 'IAC3' ||
                      p_format_list[j].mFormatID == kAudioFormat60958AC3)
                {
                    if (p_format_list[j].mSampleRate == ao->stream_format.mSampleRate)
                    {
                        i_requested_rate_format = j;
                        break;
                    }
                    if (p_format_list[j].mSampleRate == ao->sfmt_revert.mSampleRate)
                        i_current_rate_format = j;
                    else if (i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].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];
            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];
            else ao->stream_format = p_format_list[i_backup_rate_format]; /* And if we have to, any digital format will be just fine (highest rate possible). */
        }
        if (p_format_list) free(p_format_list);
    }
    if (p_streams) free(p_streams);

    if (ao->i_stream_index < 0)
    {
        ao_msg(MSGT_AO, MSGL_WARN, "can not find any digital output stream format when OpenSPDIF().\n");
        return CONTROL_FALSE;
    }

    print_format(MSGL_V, "original stream format:", &ao->sfmt_revert);

    if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
        return CONTROL_FALSE;

    err = AudioDeviceAddPropertyListener(ao->i_selected_dev,
                                         kAudioPropertyWildcardChannel,
                                         0,
                                         kAudioDevicePropertyDeviceHasChanged,
                                         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.                                     */
#ifdef WORDS_BIGENDIAN
    if (!(ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian))
#else
    if (ao->stream_format.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
        ao_msg(MSGT_AO, MSGL_WARN, "output stream has a no-native byte-order, digital output may failed.\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 + 1) * ao->chunk_size;
    ao->buffer = NULL!=ao->buffer ? realloc(ao->buffer,(ao->num_chunks + 1)*ao->chunk_size)
                                  : calloc(ao->num_chunks + 1, ao->chunk_size);

    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);


    /* Add IOProc callback. */
    err = AudioDeviceAddIOProc(ao->i_selected_dev,
                               (AudioDeviceIOProc)RenderCallbackSPDIF,
                               (void *)ao);
    if (err != noErr)
    {
        ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceAddIOProc failed: [%4.4s]\n", (char *)&err);
        return CONTROL_FALSE;
    }

    reset();

    return CONTROL_TRUE;
}

/*****************************************************************************
 * AudioDeviceSupportsDigital: Check i_dev_id for digital stream support.
 *****************************************************************************/
static int AudioDeviceSupportsDigital( AudioDeviceID i_dev_id )
{
    OSStatus                    err = noErr;
    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. */
    err = AudioDeviceGetPropertyInfo(i_dev_id, 0, FALSE,
                                     kAudioDevicePropertyStreams,
                                     &i_param_size, NULL);
    if (err != noErr)
    {
        ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
        return CONTROL_FALSE;
    }

    i_streams = i_param_size / sizeof(AudioStreamID);
    p_streams = (AudioStreamID *)malloc(i_param_size);
    if (p_streams == NULL)
    {
        ao_msg(MSGT_AO,MSGL_V, "out of memory\n");
        return CONTROL_FALSE;
    }

    err = AudioDeviceGetProperty(i_dev_id, 0, FALSE,
                                 kAudioDevicePropertyStreams,
                                 &i_param_size, p_streams);

    if (err != noErr)
    {
        ao_msg(MSGT_AO,MSGL_V, "could not get number of streams: [%4.4s]\n", (char *)&err);
        free(p_streams);
        return CONTROL_FALSE;
    }

    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 )
{
    OSStatus err = noErr;
    UInt32 i_param_size;
    AudioStreamBasicDescription *p_format_list = NULL;
    int i, i_formats, b_return = CONTROL_FALSE;

    /* Retrieve all the stream formats supported by each output stream. */
    err = AudioStreamGetPropertyInfo(i_stream_id, 0,
                                     kAudioStreamPropertyPhysicalFormats,
                                     &i_param_size, NULL);
    if (err != noErr)
    {
        ao_msg(MSGT_AO,MSGL_V, "could not get number of streamformats: [%4.4s]\n", (char *)&err);
        return CONTROL_FALSE;
    }

    i_formats = i_param_size / sizeof(AudioStreamBasicDescription);
    p_format_list = (AudioStreamBasicDescription *)malloc(i_param_size);
    if (p_format_list == NULL)
    {
        ao_msg(MSGT_AO,MSGL_V, "could not malloc the memory\n" );
        return CONTROL_FALSE;
    }

    err = AudioStreamGetProperty(i_stream_id, 0,
                                 kAudioStreamPropertyPhysicalFormats,
                                 &i_param_size, p_format_list);
    if (err != noErr)
    {
        ao_msg(MSGT_AO,MSGL_V, "could not get the list of streamformats: [%4.4s]\n", (char *)&err);
        free(p_format_list);
        return CONTROL_FALSE;
    }

    for (i = 0; i < i_formats; ++i)
    {
        print_format(MSGL_V, "supported format:", &p_format_list[i]);

        if (p_format_list[i].mFormatID == 'IAC3' ||
                  p_format_list[i].mFormatID == kAudioFormat60958AC3)
            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;
    UInt32 i_param_size = 0;
    int i;

    struct timeval now;
    struct timespec timeout;
    struct { pthread_mutex_t lock; pthread_cond_t cond; } w;

    print_format(MSGL_V, "setting stream format:", &change_format);

    /* Condition because SetProperty is asynchronious. */
    pthread_cond_init(&w.cond, NULL);
    pthread_mutex_init(&w.lock, NULL);
    pthread_mutex_lock(&w.lock);

    /* Install the callback. */
    err = AudioStreamAddPropertyListener(i_stream_id, 0,
                                         kAudioStreamPropertyPhysicalFormat,
                                         StreamListener, (void *)&w);
    if (err != noErr)
    {
        ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamAddPropertyListener failed: [%4.4s]\n", (char *)&err);
        return CONTROL_FALSE;
    }

    /* Change the format. */
    err = AudioStreamSetProperty(i_stream_id, 0, 0,
                                 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 (requiring the locks),
     * 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;

        gettimeofday(&now, NULL);
        timeout.tv_sec = now.tv_sec;
        timeout.tv_nsec = (now.tv_usec + 500000) * 1000;

        if (pthread_cond_timedwait(&w.cond, &w.lock, &timeout))
            ao_msg(MSGT_AO, MSGL_V, "reached timeout\n" );

        i_param_size = sizeof(AudioStreamBasicDescription);
        err = AudioStreamGetProperty(i_stream_id, 0,
                                     kAudioStreamPropertyPhysicalFormat,
                                     &i_param_size,
                                     &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 = AudioStreamRemovePropertyListener(i_stream_id, 0,
                                            kAudioStreamPropertyPhysicalFormat,
                                            StreamListener);
    if (err != noErr)
    {
        ao_msg(MSGT_AO, MSGL_WARN, "AudioStreamRemovePropertyListener failed: [%4.4s]\n", (char *)&err);
        return CONTROL_FALSE;
    }

    /* Destroy the lock and condition. */
    pthread_mutex_unlock(&w.lock);
    pthread_mutex_destroy(&w.lock);
    pthread_cond_destroy(&w.cond);

    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 = buf_used();
    int req = outOutputData->mBuffers[ao->i_stream_index].mDataByteSize;

    if (amt > req)
        amt = req;
    if (amt)
        read_buffer((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;

    // 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 support digital format output, let's set it. */
            ao_msg(MSGT_AO, MSGL_V, "detected current stream support digital, try to restore digital output...\n");

            if (!AudioStreamChangeFormat(ao->i_stream_id, ao->stream_format))
            {
                ao_msg(MSGT_AO, MSGL_WARN, "restore digital output failed.\n");
            }
            else
            {
                ao_msg(MSGT_AO, MSGL_WARN, "restore digital output succeed.\n");
                reset();
            }
        }
        else
            ao_msg(MSGT_AO, MSGL_V, "detected current stream do not support digital.\n");
    }

    wrote=write_buffer(output_samples, num_bytes);
    audio_resume();
    return wrote;
}

/* set variables and buffer to initial state */
static void reset(void)
{
  audio_pause();
  /* reset ring-buffer state */
  ao->buf_read_pos=0;
  ao->buf_write_pos=0;
  
  return;
}


/* return available space */
static int get_space(void)
{
  return buf_free();
}


/* return delay until audio is played */
static float get_delay(void)
{
  int buffered = ao->buffer_len - ao->chunk_size - buf_free(); // could be less
  // inaccurate, should also contain the data buffered e.g. by the OS
  return (float)(buffered)/(float)ao_data.bps;
}


/* unload plugin and deregister from coreaudio */
static void uninit(int immed)
{
  OSStatus            err = noErr;
  UInt32              i_param_size = 0;

  if (!immed) {
    long long timeleft=(1000000LL*buf_used())/ao_data.bps;
    ao_msg(MSGT_AO,MSGL_DBG2, "%d bytes left @%d bps (%d usec)\n", buf_used(), ao_data.bps, (int)timeleft);
    usec_sleep((int)timeleft);
  }

  if (!ao->b_digital) {
      AudioOutputUnitStop(ao->theOutputUnit);
      AudioUnitUninitialize(ao->theOutputUnit);
      CloseComponent(ao->theOutputUnit);
  }
  else {
      /* Stop device. */
      err = AudioDeviceStop(ao->i_selected_dev,
                            (AudioDeviceIOProc)RenderCallbackSPDIF);
      if (err != noErr)
          ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStop failed: [%4.4s]\n", (char *)&err);

      /* Remove IOProc callback. */
      err = AudioDeviceRemoveIOProc(ao->i_selected_dev,
                                    (AudioDeviceIOProc)RenderCallbackSPDIF);
      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)
      {
          int b_mix;
          Boolean b_writeable;
          /* Revert mixable to true if we are allowed to. */
          err = AudioDeviceGetPropertyInfo(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
                                           &i_param_size, &b_writeable);
          err = AudioDeviceGetProperty(ao->i_selected_dev, 0, FALSE, kAudioDevicePropertySupportsMixing,
                                       &i_param_size, &b_mix);
          if (err != noErr && b_writeable)
          {
              b_mix = 1;
              err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
                                           kAudioDevicePropertySupportsMixing, i_param_size, &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;
          i_param_size = sizeof(ao->i_hog_pid);
          err = AudioDeviceSetProperty(ao->i_selected_dev, 0, 0, FALSE,
                                       kAudioDevicePropertyHogMode, i_param_size, &ao->i_hog_pid);
          if (err != noErr) ao_msg(MSGT_AO, MSGL_WARN, "Could not release hogmode: [%4.4s]\n", (char *)&err);
      }
  }

  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, (AudioDeviceIOProc)RenderCallbackSPDIF);
        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, (AudioDeviceIOProc)RenderCallbackSPDIF);
        if (err != noErr)
            ao_msg(MSGT_AO, MSGL_WARN, "AudioDeviceStart failed: [%4.4s]\n", (char *)&err);
    }
    ao->paused = 0;
}

/*****************************************************************************
 * StreamListener
 *****************************************************************************/
static OSStatus StreamListener( AudioStreamID inStream,
                                UInt32 inChannel,
                                AudioDevicePropertyID inPropertyID,
                                void * inClientData )
{
    struct { pthread_mutex_t lock; pthread_cond_t cond; } * w = inClientData;

    switch (inPropertyID)
    {
        case kAudioStreamPropertyPhysicalFormat:
            if (NULL!=w)
            {
                pthread_mutex_lock(&w->lock);
                pthread_cond_signal(&w->cond);
                pthread_mutex_unlock(&w->lock);
            }
        default:
            break;
    }
    return noErr;
}

static OSStatus DeviceListener( AudioDeviceID inDevice,
                                UInt32 inChannel,
                                Boolean isInput,
                                AudioDevicePropertyID inPropertyID,
                                void* inClientData )
{
    switch (inPropertyID)
    {
        case kAudioDevicePropertyDeviceHasChanged:
            ao_msg(MSGT_AO, MSGL_WARN, "got notify kAudioDevicePropertyDeviceHasChanged.\n");
            ao->b_stream_format_changed = 1;
        default:
            break;
    }
    return noErr;
}