qtsound: added audio capture functionality for MAC OS X
authorMichael Feurstein <michael.feurstein@gmail.com>
Sun, 30 Oct 2011 21:27:46 +0000 (22:27 +0100)
committerJean-Baptiste Kempf <jb@videolan.org>
Fri, 4 Nov 2011 01:16:24 +0000 (02:16 +0100)
Tested with Built-in Input, Built-in Microphone and Griffin iMic USB system
Usage: qtsound://"Built-In Input" or qtsound://"iMic USB audio system"

Modified-by: Jean-Baptiste Kempf <jb@videolan.org>
Signed-off-by: Jean-Baptiste Kempf <jb@videolan.org>
configure.ac
modules/access/Modules.am
modules/access/qtsound.m [new file with mode: 0644]

index 8f631c4..9db6f6a 100644 (file)
@@ -2101,19 +2101,16 @@ then
 fi
 
 dnl
-dnl QTCapture
-AC_ARG_ENABLE(macosx-qtcapture,
-  [  --enable-macosx-qtcapture Mac OS X qtcapture (iSight) module (default enabled on Mac OS X)])
-if test "x${enable_macosx_qtcapture}" != "xno" &&
-  (test "${SYS}" = "darwin" || test "${enable_macosx_qtcapture}" = "yes")
+dnl QTKit
+AC_ARG_ENABLE(macosx-qtkit,
+  [  --enable-macosx-qtkit Mac OS X qtcapture (video) and qtsound (audio) module (default enabled on Mac OS X)])
+if test "x${enable_macosx_qtkit}" != "xno" &&
+  (test "${SYS}" = "darwin" || test "${enable_macosx_qtkit}" = "yes")
 then
-  VLC_ADD_LIBS([qtcapture], [-Wl,-framework,Cocoa])
-  VLC_ADD_LIBS([qtcapture], [-Wl,-framework,QTKit])
-  VLC_ADD_LIBS([qtcapture], [-Wl,-framework,CoreAudio])
-  VLC_ADD_LIBS([qtcapture], [-Wl,-framework,QuartzCore])
-  VLC_ADD_LIBS([qtcapture], [-Wl,-framework,CoreVideo])
   VLC_ADD_PLUGIN([qtcapture])
+  VLC_ADD_PLUGIN([qtsound])
 fi
+AM_CONDITIONAL(HAVE_QTKIT, [test "${SYS}" = "darwin" && "x${enable_macosx_qtkit}" != "xno"])
 
 
 dnl
index 45b6499..81d8e78 100644 (file)
@@ -55,7 +55,6 @@ SOURCES_dvdnav = dvdnav.c
 SOURCES_dvdread = dvdread.c
 SOURCES_dc1394 = dc1394.c
 SOURCES_pvr = pvr.c
-SOURCES_qtcapture = qtcapture.m
 SOURCES_linsys_sdi = linsys/linsys_sdi.c linsys/linsys_sdi.h
 SOURCES_linsys_hdsdi = \
        linsys/linsys_hdsdi.c \
@@ -83,6 +82,23 @@ SOURCES_htcpcp = htcpcp.c
 SOURCES_access_rar = rar/rar.c rar/rar.h rar/access.c
 SOURCES_stream_filter_rar = rar/rar.c rar/rar.h rar/stream.c
 
+if HAVE_QTKIT
+libqtcapture_plugin_la_SOURCES = qtcapture.m
+libqtcapture_plugin_la_LIBADDD = $(AM_LIBADD)
+libqtcapture_plugin_la_LDFLAGS = -Wl,-framework,Cocoa -Wl,-framework,QTKit -Wl,-framework,CoreVideo -Wl,-framework,QuartzCore
+libqtcapture_plugin_la_DEPENDENCIES =
+
+libqtsound_plugin_la_SOURCES = qtsound.m
+libqtsound_plugin_la_LIBADD = $(AM_LIBADD)
+libqtsound_plugin_la_LDFLAGS = -Wl,-framework,Cocoa -Wl,-framework,QTKit -Wl,-framework,CoreAudio
+libqtsound_plugin_la_DEPENDENCIES =
+
+libvlc_LTLIBRARIES += \
+        libqtcapture_plugin.la \
+        libqtsound_plugin.la \
+        $(NULL)
+endif
+
 libaccess_rtmp_plugin_la_SOURCES = \
         rtmp/access.c \
         rtmp/rtmp_amf_flv.c \
diff --git a/modules/access/qtsound.m b/modules/access/qtsound.m
new file mode 100644 (file)
index 0000000..75a8ef7
--- /dev/null
@@ -0,0 +1,606 @@
+/*****************************************************************************
+* qtsound.m: qtkit (Mac OS X) based audio capture module
+*****************************************************************************
+* Copyright © 2011 VLC authors and VideoLAN
+*
+* Authors: Pierre d'Herbemont <pdherbemont@videolan.org>
+*          Gustaf Neumann <neumann@wu.ac.at>
+*          Michael S. Feurstein <michael.feurstein@wu.ac.at>
+*
+*****************************************************************************
+* This library is free software; you can redistribute it and/or
+* modify it under the terms of the GNU Lesser General Public License
+* as published by the Free Software Foundation; either version 2.1
+* of the License, or (at your option) any later version.
+*
+* This library 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
+* Lesser General Public License for more details.
+*
+* You should have received a copy of the GNU Lesser General Public
+* License along with this library; if not, write to the Free Software
+* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110, USA
+*
+*****************************************************************************/
+
+/*****************************************************************************
+ * Preamble
+ *****************************************************************************/
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <vlc_common.h>
+#include <vlc_plugin.h>
+#include <vlc_aout.h>
+
+#include <vlc_demux.h>
+#include <vlc_dialog.h>
+
+#import <QTKit/QTKit.h>
+
+/*****************************************************************************
+ * Local prototypes.
+ *****************************************************************************/
+static int Open( vlc_object_t *p_this );
+static void Close( vlc_object_t *p_this );
+static int Demux( demux_t *p_demux );
+static int Control( demux_t *, int, va_list );
+
+/*****************************************************************************
+ * Module descriptor
+ *****************************************************************************/
+
+vlc_module_begin()
+set_shortname( N_("QTSound") )
+set_description( N_("QuickTime Sound Capture") )
+set_category( CAT_INPUT )
+set_subcategory( SUBCAT_INPUT_ACCESS )
+add_shortcut( "qtsound" )
+set_capability( "access_demux", 0 )
+set_callbacks( Open, Close )
+vlc_module_end ()
+
+
+/*****************************************************************************
+ * QTKit Bridge
+ *****************************************************************************/
+@interface VLCDecompressedAudioOutput : QTCaptureDecompressedAudioOutput
+{
+    demux_t *p_qtsound;
+    AudioBuffer *currentAudioBuffer;
+    UInt32 numberOfSamples;
+    date_t date;
+    mtime_t currentPts;
+    mtime_t previousPts;
+}
+- (id)initWithDemux:(demux_t *)p_demux;
+- (void)outputAudioSampleBuffer:(QTSampleBuffer *)sampleBuffer fromConnection:(QTCaptureConnection *)connection;
+- (BOOL)checkCurrentAudioBuffer;
+- (mtime_t)getCurrentPts;
+- (void *)getCurrentAudioBufferData;
+- (UInt32)getCurrentTotalDataSize;
+- (UInt32)getNumberOfSamples;
+
+@end
+
+@implementation VLCDecompressedAudioOutput : QTCaptureDecompressedAudioOutput
+- (id)initWithDemux:(demux_t *)p_demux
+{
+    if( self = [super init] )
+    {
+        p_qtsound = p_demux;
+        currentAudioBuffer = nil;
+        date_Init(&date, 44100, 1);
+        date_Set(&date,0);
+        currentPts = 0;
+        previousPts = 0;
+    }
+    return self;
+}
+- (void)dealloc
+{
+    @synchronized (self)
+    {
+        free(currentAudioBuffer);
+        currentAudioBuffer = nil;
+    }
+    [super dealloc];
+}
+
+- (void)outputAudioSampleBuffer:(QTSampleBuffer *)sampleBuffer fromConnection:(QTCaptureConnection *)connection
+{
+    AudioBufferList *tempAudioBufferList;
+    block_t *rawAudioData;
+    UInt32 totalDataSize = 0;
+    UInt32 count = 0;
+
+    @synchronized (self)
+    {
+        numberOfSamples = [sampleBuffer numberOfSamples];
+        date_Increment(&date,numberOfSamples);
+        currentPts = date_Get(&date);
+
+        tempAudioBufferList = [sampleBuffer audioBufferListWithOptions:0];
+        if (tempAudioBufferList->mNumberBuffers == 2)
+        {
+            /*
+             * Compute totalDataSize as sum of all data blocks in the
+             * audio buffer list:
+             */
+            for ( count = 0; count < tempAudioBufferList->mNumberBuffers; count++ )
+            {
+                totalDataSize += tempAudioBufferList->mBuffers[count].mDataByteSize;
+            }
+            /*
+             * Allocate storage for the interleaved audio data
+             */
+            rawAudioData = block_Alloc(totalDataSize * sizeof(float));
+            if (NULL == rawAudioData)
+            {
+                msg_Err( p_qtsound, "Raw audiodata could not be allocated" );
+                return;
+            }
+        }
+        else
+        {
+            msg_Err( p_qtsound, "Too many or only one channel found." );
+            return;
+        }
+
+        /*
+         * Interleave raw data (provided in two separate channels as
+         * F32L) with 2 samples per frame
+         */
+        if ( totalDataSize )
+        {
+            unsigned short i;
+            const float *b1Ptr, *b2Ptr;
+            float *uPtr;
+
+            for (i = 0,
+                 uPtr = (float *)rawAudioData,
+                 b1Ptr = (const float *) tempAudioBufferList->mBuffers[0].mData,
+                 b2Ptr = (const float *) tempAudioBufferList->mBuffers[1].mData;
+                 i < numberOfSamples; i++)
+            {
+                *uPtr = *b1Ptr;
+                uPtr ++;
+                b1Ptr ++;
+                *uPtr = *b2Ptr;
+                uPtr ++;
+                b2Ptr ++;
+            }
+
+            if (currentAudioBuffer == nil)
+            {
+                currentAudioBuffer = (AudioBuffer *)malloc(sizeof(AudioBuffer));
+                if (NULL == currentAudioBuffer)
+                {
+                    msg_Err( p_qtsound, "AudioBuffer could not be allocated." );
+                    return;
+                }
+            }
+            currentAudioBuffer->mNumberChannels = 2;
+            currentAudioBuffer->mDataByteSize = totalDataSize;
+            currentAudioBuffer->mData = rawAudioData;
+        }
+        free(rawAudioData);
+    }
+}
+
+- (BOOL)checkCurrentAudioBuffer
+{
+    return (currentAudioBuffer) ? 1 : 0;
+}
+
+- (mtime_t)getCurrentPts
+{
+    /* FIXME: can this getter be minimized? */
+    mtime_t pts;
+
+    if( !currentAudioBuffer || currentPts == previousPts )
+    {
+        return 0;
+    }
+
+    @synchronized (self)
+    {
+        pts = previousPts = currentPts;
+    }
+
+    return (currentAudioBuffer->mData) ? currentPts : 0;
+}
+
+- (void *)getCurrentAudioBufferData
+{
+    return currentAudioBuffer->mData;
+}
+
+- (UInt32)getCurrentTotalDataSize
+{
+    return currentAudioBuffer->mDataByteSize;
+}
+
+- (UInt32)getNumberOfSamples
+{
+    return numberOfSamples;
+}
+
+@end
+
+/*****************************************************************************
+ * Struct
+ *****************************************************************************/
+
+struct demux_sys_t {
+    QTCaptureSession * session;
+    QTCaptureDevice * audiodevice;
+    VLCDecompressedAudioOutput * audiooutput;
+    es_out_id_t *p_es_audio;
+    int i_audio_max_buffer_size;
+};
+
+/*****************************************************************************
+ * Open: initialize interface
+ *****************************************************************************/
+static int Open( vlc_object_t *p_this )
+{
+    demux_t *p_demux = (demux_t*)p_this;
+    demux_sys_t *p_sys;
+    es_format_t audiofmt;
+    char *psz_uid = NULL;
+    int audiocodec;
+    bool success;
+    NSString *qtk_curraudiodevice_uid;
+    NSAutoreleasePool *pool;
+    NSArray *myAudioDevices, *audioformat_array;
+    QTFormatDescription *audio_format;
+    QTCaptureDeviceInput *audioInput;
+    NSError *o_returnedAudioError;
+
+    if( p_demux->psz_location && *p_demux->psz_location )
+    {
+        psz_uid = p_demux->psz_location;
+    }
+    msg_Dbg( p_demux, "qtsound uid = %s", psz_uid );
+    qtk_curraudiodevice_uid = [[NSString alloc] initWithFormat:@"%s", psz_uid];
+
+    pool = [[NSAutoreleasePool alloc] init];
+
+    p_demux->p_sys = p_sys = calloc( 1, sizeof( demux_sys_t ) );
+    if( !p_sys )
+        return VLC_ENOMEM;
+
+    msg_Dbg( p_demux, "qtsound : uid = %s", [qtk_curraudiodevice_uid UTF8String]);
+    myAudioDevices = [[[QTCaptureDevice inputDevicesWithMediaType:QTMediaTypeSound] arrayByAddingObjectsFromArray:[QTCaptureDevice inputDevicesWithMediaType:QTMediaTypeMuxed]] retain];
+    if([myAudioDevices count] == 0)
+    {
+        dialog_FatalWait( p_demux, _("No Audio Input device found"),
+                         _("Your Mac does not seem to be equipped with a suitable audio input device."
+                     "Please check your connectors and drivers.") );
+        msg_Err( p_demux, "Can't find any Audio device" );
+
+        goto error;
+    }
+    int iaudio;
+    for(iaudio = 0; iaudio < [myAudioDevices count]; iaudio++){
+        QTCaptureDevice *qtk_audioDevice;
+        qtk_audioDevice = [myAudioDevices objectAtIndex:iaudio];
+        msg_Dbg( p_demux, "qtsound audio %d/%d localizedDisplayName: %s uniqueID: %s", iaudio, [myAudioDevices count], [[qtk_audioDevice localizedDisplayName] UTF8String], [[qtk_audioDevice uniqueID] UTF8String]);
+        if([[[qtk_audioDevice localizedDisplayName]stringByTrimmingCharactersInSet:[NSCharacterSet whitespaceCharacterSet]] isEqualToString:qtk_curraudiodevice_uid]){
+            msg_Dbg( p_demux, "Device found" );
+            break;
+        }
+    }
+
+    audioInput = nil;
+    if(iaudio < [myAudioDevices count]){
+        p_sys->audiodevice = [myAudioDevices objectAtIndex:iaudio];
+    }
+    else
+    {
+        /* cannot find designated audio device, fall back to open default audio device */
+        msg_Dbg(p_demux, "Cannot find designated uid audio device as %s. Fall back to open default audio device.", [qtk_curraudiodevice_uid UTF8String]);
+        p_sys->audiodevice = [QTCaptureDevice defaultInputDeviceWithMediaType: QTMediaTypeSound];
+    }
+    if( !p_sys->audiodevice )
+    {
+        dialog_FatalWait( p_demux, _("No audio input device found"),
+                         _("Your Mac does not seem to be equipped with a suitable audio input device."
+                     "Please check your connectors and drivers.") );
+        msg_Err( p_demux, "Can't find any Audio device" );
+
+        goto error;
+    }
+
+    if( ![p_sys->audiodevice open: &o_returnedAudioError] )
+    {
+        msg_Err( p_demux, "Unable to open the audio capture device (%d)", [o_returnedAudioError code] );
+        goto error;
+    }
+
+    if( [p_sys->audiodevice isInUseByAnotherApplication] == YES )
+    {
+        msg_Err( p_demux, "default audio capture device is exclusively in use by another application" );
+        goto error;
+    }
+    audioInput = [[QTCaptureDeviceInput alloc] initWithDevice: p_sys->audiodevice];
+    if( !audioInput )
+    {
+        msg_Err( p_demux, "can't create a valid audio capture input facility" );
+        goto error;
+    } else {
+        msg_Dbg( p_demux, "created valid audio capture input facility" );
+    }
+
+    p_sys->audiooutput = [[VLCDecompressedAudioOutput alloc] initWithDemux:p_demux];
+    msg_Dbg ( p_demux, "initialized audio output" );
+
+    /* Get the formats */
+    /*
+     FIXME: the format description gathered here does not seem to be the same
+     in comparison to the format description collected from the actual sampleBuffer.
+     This information needs to be updated some other place. For the time being this shall suffice.
+
+     The following verbose output is an example of what is read from the input device during the below block
+     [0x3042138] qtsound demux debug: Audio localized format summary: Linear PCM, 24 bit little-endian signed integer, 2 channels, 44100 Hz
+     [0x3042138] qtsound demux debug: Sample Rate: 44100; Format ID: lpcm; Format Flags: 00000004; Bytes per Packet: 8; Frames per Packet: 1; Bytes per Frame: 8; Channels per Frame: 2; Bits per Channel: 24
+     [0x3042138] qtsound demux debug: Flag float 0 bigEndian 0 signedInt 1 packed 0 alignedHigh 0 non interleaved 0 non mixable 0
+     canonical 0 nativeFloatPacked 0 nativeEndian 0
+
+     However when reading this information from the sampleBuffer during the delegate call from
+     - (void)outputAudioSampleBuffer:(QTSampleBuffer *)sampleBuffer fromConnection:(QTCaptureConnection *)connection;
+     the following data shows up
+     2011-09-23 22:06:03.077 VLC[23070:f103] Audio localized format summary: Linear PCM, 32 bit little-endian floating point, 2 channels, 44100 Hz
+     2011-09-23 22:06:03.078 VLC[23070:f103] Sample Rate: 44100; Format ID: lpcm; Format Flags: 00000029; Bytes per Packet: 4; Frames per Packet: 1; Bytes per Frame: 4; Channels per Frame: 2; Bits per Channel: 32
+     2011-09-23 22:06:03.078 VLC[23070:f103] Flag float 1 bigEndian 0 signedInt 0 packed 1 alignedHigh 0 non interleaved 1 non mixable 0
+     canonical 1 nativeFloatPacked 1 nativeEndian 0
+
+     Note the differences
+     24bit vs. 32bit
+     little-endian signed integer vs. little-endian floating point
+     format flag 00000004 vs. 00000029
+     bytes per packet 8 vs. 4
+     packed 0 vs. 1
+     non interleaved 0 vs. 1 -> this makes a major difference when filling our own buffer
+     canonical 0 vs. 1
+     nativeFloatPacked 0 vs. 1
+
+     One would assume we'd need to feed the (es_format_t)audiofmt with the data collected here.
+     This is not the case. Audio will be transmitted in artefacts, due to wrong information.
+
+     At the moment this data is set manually, however one should consider trying to set this data dynamically
+     */
+    audioformat_array = [p_sys->audiodevice formatDescriptions];
+    audio_format = NULL;
+    for( int k = 0; k < [audioformat_array count]; k++ )
+    {
+        audio_format = (QTFormatDescription *)[audioformat_array objectAtIndex: k];
+
+        msg_Dbg( p_demux, "Audio localized format summary: %s", [[audio_format localizedFormatSummary] UTF8String]);
+        msg_Dbg( p_demux, "Audio format description attributes: %s",[[[audio_format formatDescriptionAttributes] description] UTF8String]);
+
+        AudioStreamBasicDescription asbd = {0};
+        NSValue *asbdValue =  [audio_format attributeForKey:QTFormatDescriptionAudioStreamBasicDescriptionAttribute];
+        [asbdValue getValue:&asbd];
+
+        char formatIDString[5];
+        UInt32 formatID = CFSwapInt32HostToBig (asbd.mFormatID);
+        bcopy (&formatID, formatIDString, 4);
+        formatIDString[4] = '\0';
+
+        /* kept for development purposes */
+#if 0
+        msg_Dbg( p_demux, "Sample Rate: %.0lf; Format ID: %s; Format Flags: %.8x; Bytes per Packet: %d; Frames per Packet: %d; Bytes per Frame: %d; Channels per Frame: %d; Bits per Channel: %d",
+                asbd.mSampleRate,
+                formatIDString,
+                asbd.mFormatFlags,
+                asbd.mBytesPerPacket,
+                asbd.mFramesPerPacket,
+                asbd.mBytesPerFrame,
+                asbd.mChannelsPerFrame,
+                asbd.mBitsPerChannel);
+
+        msg_Dbg( p_demux, "Flag float %d bigEndian %d signedInt %d packed %d alignedHigh %d non interleaved %d non mixable %d\ncanonical %d nativeFloatPacked %d nativeEndian %d",
+                (asbd.mFormatFlags & kAudioFormatFlagIsFloat) != 0,
+                (asbd.mFormatFlags & kAudioFormatFlagIsBigEndian) != 0,
+                (asbd.mFormatFlags & kAudioFormatFlagIsSignedInteger) != 0,
+                (asbd.mFormatFlags & kAudioFormatFlagIsPacked) != 0,
+                (asbd.mFormatFlags & kAudioFormatFlagIsAlignedHigh) != 0,
+                (asbd.mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0,
+                (asbd.mFormatFlags & kAudioFormatFlagIsNonMixable) != 0,
+
+                (asbd.mFormatFlags & kAudioFormatFlagsCanonical) != 0,
+                (asbd.mFormatFlags & kAudioFormatFlagsNativeFloatPacked) != 0,
+                (asbd.mFormatFlags & kAudioFormatFlagsNativeEndian) != 0
+                );
+#endif
+    }
+
+    if( [audioformat_array count] )
+        audio_format = [audioformat_array objectAtIndex: 0];
+    else goto error;
+
+    /* Now we can init */
+    audiocodec = VLC_CODEC_FL32;
+    es_format_Init( &audiofmt, AUDIO_ES, audiocodec);
+
+    audiofmt.audio.i_format = audiocodec;
+    audiofmt.audio.i_rate = 44100;
+    /*
+     * i_physical_channels Describes the channels configuration of the
+     * samples (ie. number of channels which are available in the
+     * buffer, and positions).
+     */
+    audiofmt.audio.i_physical_channels = AOUT_CHAN_RIGHT | AOUT_CHAN_LEFT;
+    /*
+     * i_original_channels Describes from which original channels,
+     * before downmixing, the buffer is derived.
+     */
+    audiofmt.audio.i_original_channels = AOUT_CHAN_RIGHT | AOUT_CHAN_LEFT;
+    /*
+     * i_bytes_per_frame Optional - for A/52, SPDIF and DTS types:
+     * Bytes used by one compressed frame, depends on bitrate.
+     */
+    audiofmt.audio.i_bytes_per_frame = 4;
+    /*
+     * Number of sampleframes contained in one compressed frame.
+     */
+    audiofmt.audio.i_frame_length = 1;
+    /*
+     * Please note that it may be completely arbitrary - buffers are not
+     * obliged to contain a integral number of so-called "frames". It's
+     * just here for the division:
+     * buffer_size = i_nb_samples * i_bytes_per_frame / i_frame_length
+     */
+    audiofmt.audio.i_bitspersample = 32;
+    audiofmt.audio.i_channels = 2;
+    audiofmt.audio.i_blockalign = audiofmt.audio.i_channels * audiofmt.audio.i_bitspersample / 16;
+    audiofmt.i_bitrate = audiofmt.audio.i_channels * audiofmt.audio.i_rate * audiofmt.audio.i_bitspersample;
+    p_sys->i_audio_max_buffer_size = 4096;
+
+    p_sys->session = [[QTCaptureSession alloc] init];
+
+    success = [p_sys->session addInput:audioInput error: &o_returnedAudioError];
+    if( !success )
+    {
+        msg_Err( p_demux, "the audio capture device could not be added to capture session (%d)", [o_returnedAudioError code] );
+        goto error;
+    }
+
+    success = [p_sys->session addOutput:p_sys->audiooutput error: &o_returnedAudioError];
+    if( !success )
+    {
+        msg_Err( p_demux, "audio output could not be added to capture session (%d)", [o_returnedAudioError code] );
+        goto error;
+    }
+
+    [p_sys->session startRunning];
+
+    /* Set up p_demux */
+    p_demux->pf_demux = Demux;
+    p_demux->pf_control = Control;
+    p_demux->info.i_update = 0;
+    p_demux->info.i_title = 0;
+    p_demux->info.i_seekpoint = 0;
+
+    msg_Dbg( p_demux, "New audio es %d channels %dHz",
+            audiofmt.audio.i_channels, audiofmt.audio.i_rate );
+
+    p_sys->p_es_audio = es_out_Add( p_demux->out, &audiofmt );
+
+    [audioInput release];
+    [pool release];
+
+    msg_Dbg( p_demux, "QTSound: We have an audio device ready!" );
+
+    return VLC_SUCCESS;
+error:
+    [audioInput release];
+    [pool release];
+
+    free( p_sys );
+
+    return VLC_EGENERIC;
+}
+
+/*****************************************************************************
+ * Close: destroy interface
+ *****************************************************************************/
+static void Close( vlc_object_t *p_this )
+{
+    NSAutoreleasePool *pool = [[NSAutoreleasePool alloc] init];
+    demux_t *p_demux = (demux_t*)p_this;
+    demux_sys_t *p_sys = p_demux->p_sys;
+
+    [p_sys->session performSelectorOnMainThread:@selector(stopRunning) withObject:nil waitUntilDone:NO];
+    [p_sys->audiooutput performSelectorOnMainThread:@selector(release) withObject:nil waitUntilDone:NO];
+    [p_sys->session performSelectorOnMainThread:@selector(release) withObject:nil waitUntilDone:NO];
+
+    free( p_sys );
+
+    [pool release];
+}
+
+/*****************************************************************************
+ * Demux:
+ *****************************************************************************/
+static int Demux( demux_t *p_demux )
+{
+    demux_sys_t *p_sys = p_demux->p_sys;
+    block_t *p_blocka;
+    NSAutoreleasePool *pool;
+
+    p_blocka = block_New( p_demux, p_sys->i_audio_max_buffer_size );
+
+    if( !p_blocka )
+    {
+        msg_Err( p_demux, "cannot get audio block" );
+        return 0;
+    }
+
+    pool = [[NSAutoreleasePool alloc] init];
+
+    @synchronized (p_sys->audiooutput)
+    {
+        if ( [p_sys->audiooutput checkCurrentAudioBuffer] )
+        {
+            p_blocka->i_pts = [p_sys->audiooutput getCurrentPts];
+            p_blocka->p_buffer = [p_sys->audiooutput getCurrentAudioBufferData];
+            p_blocka->i_nb_samples = [p_sys->audiooutput getNumberOfSamples];
+            p_blocka->i_buffer = [p_sys->audiooutput getCurrentTotalDataSize];
+        }
+    }
+
+    if( !p_blocka->i_pts )
+    {
+        // Nothing to transfer yet, just forget
+        block_Release( p_blocka );
+        [pool release];
+        msleep( 10000 );
+        return 1;
+    }
+
+    [pool release];
+
+    if( p_blocka )
+    {
+        es_out_Control( p_demux->out, ES_OUT_SET_PCR, p_blocka->i_pts );
+        es_out_Send( p_demux->out, p_sys->p_es_audio, p_blocka );
+    }
+
+    return 1;
+}
+
+/*****************************************************************************
+ * Control:
+ *****************************************************************************/
+static int Control( demux_t *p_demux, int i_query, va_list args )
+{
+    bool *pb;
+    int64_t *pi64;
+
+    switch( i_query )
+    {
+            /* Special for access_demux */
+        case DEMUX_CAN_PAUSE:
+        case DEMUX_CAN_SEEK:
+        case DEMUX_SET_PAUSE_STATE:
+        case DEMUX_CAN_CONTROL_PACE:
+            pb = (bool*)va_arg( args, bool * );
+            *pb = false;
+            return VLC_SUCCESS;
+
+        case DEMUX_GET_PTS_DELAY:
+            pi64 = (int64_t*)va_arg( args, int64_t * );
+            *pi64 = INT64_C(1000) * var_InheritInteger( p_demux, "live-caching" );
+            return VLC_SUCCESS;
+
+        default:
+            return VLC_EGENERIC;
+    }
+    return VLC_EGENERIC;
+}