view libao2/ao_coreaudio.c @ 29946:54bacf6a38ca

Update the SuperH VEU vidix driver with code that calls fsync() after each frame to make sure the frame is flushed in case of deferred io. Patch by Magnus Damm, damm opensource se

/*
 * 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 "mp_msg.h"

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

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? */

  /* 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;
  av_fifo_generic_read(ao->buffer, data, len, NULL);
  return len;
}

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.
			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:
		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 %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(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( 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;
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);

    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;

    /* 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);
            goto err_out;
        }

        /* 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);
            goto err_out;
        }

        /* 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);
            free( psz_name);
            goto err_out;
        }

        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);
		goto err_out;
	}

    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.
#if HAVE_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" );
            goto err_out;
        }
        ao->stream_format = inDesc;
        return OpenSPDIF();
    }

	/* original analog output code */
	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");
		goto err_out;
	}

	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);
		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;
	}

	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:
    CloseComponent(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;

    /* 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);
        ao->i_hog_pid = -1;
        goto err_out;
    }

    /* 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);
        goto err_out;
    }

    /* 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);
        goto err_out;
    }

    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" );
        goto err_out;
    }

    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);
        goto err_out;
    }

    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");
        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;

    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.                                     */
#if HAVE_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 * 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);


    /* 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);
        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 = 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);
    }
    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 )
{
    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;

    static volatile int stream_format_changed;
    stream_format_changed = 0;

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

    /* Install the callback. */
    err = AudioStreamAddPropertyListener(i_stream_id, 0,
                                         kAudioStreamPropertyPhysicalFormat,
                                         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 = 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,
     * 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" );

        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;
    }

    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;

    // 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();
  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;
  UInt32              i_param_size = 0;

  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);
      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);
      }
  }

  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, (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 )
{
    switch (inPropertyID)
    {
        case kAudioStreamPropertyPhysicalFormat:
            ao_msg(MSGT_AO, MSGL_V, "got notify kAudioStreamPropertyPhysicalFormat changed.\n");
            if (inClientData)
                *(volatile int *)inClientData = 1;
        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;
}