mpv/libao2/ao_macosx.c

/*
 *
 *  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 <sys/time.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;
}