Skip to content

V
vlc
Commit b8b5a70c authored by Felix Paul Kühne's avatar Felix Paul Kühne
Browse files
Options

qtsound: drop legacy NSAutoreleasePool pattern

parent 210bdda1
Hide whitespace changes
Inline Side-by-side
Showing with 228 additions and 228 deletions
modules/access/qtsound.m
View file @ b8b5a70c
......@@ -239,235 +239,237 @@ static int Open(vlc_object_t *p_this)
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;
@autoreleasepool {
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];
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;
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");
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;
}
unsigned iaudio;
for (iaudio = 0; iaudio < [myAudioDevices count]; iaudio++) {
QTCaptureDevice *qtk_audioDevice;
qtk_audioDevice = [myAudioDevices objectAtIndex:iaudio];
msg_Dbg(p_demux, "qtsound audio %u/%lu localizedDisplayName: %s uniqueID: %s", iaudio, [myAudioDevices count], [[qtk_audioDevice localizedDisplayName] UTF8String], [[qtk_audioDevice uniqueID] UTF8String]);
if ([[[qtk_audioDevice uniqueID]stringByTrimmingCharactersInSet:[NSCharacterSet whitespaceCharacterSet]] isEqualToString:qtk_curraudiodevice_uid]) {
msg_Dbg(p_demux, "Device found");
break;
goto error;
}
unsigned iaudio;
for (iaudio = 0; iaudio < [myAudioDevices count]; iaudio++) {
QTCaptureDevice *qtk_audioDevice;
qtk_audioDevice = [myAudioDevices objectAtIndex:iaudio];
msg_Dbg(p_demux, "qtsound audio %u/%lu localizedDisplayName: %s uniqueID: %s",
iaudio, [myAudioDevices count],
[[qtk_audioDevice localizedDisplayName] UTF8String],
[[qtk_audioDevice uniqueID] UTF8String]);
if ([[[qtk_audioDevice uniqueID] 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");
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;
}
goto error;
}
if(![p_sys->audiodevice open: &o_returnedAudioError]) {
msg_Err(p_demux, "Unable to open the audio capture device (%ld)", [o_returnedAudioError code]);
goto error;
}
if(![p_sys->audiodevice open: &o_returnedAudioError]) {
msg_Err(p_demux, "Unable to open the audio capture device (%ld)", [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([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");
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;
/*
* 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 / 8);
audiofmt.i_bitrate = audiofmt.audio.i_channels * audiofmt.audio.i_rate * audiofmt.audio.i_bitspersample;
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 (%ld)", [o_returnedAudioError code]);
goto error;
}
p_sys->audiooutput = [[VLCDecompressedAudioOutput alloc] initWithDemux:p_demux];
msg_Dbg (p_demux, "initialized audio output");
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 (%ld)", [o_returnedAudioError code]);
goto error;
}
/* 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];
[p_sys->session startRunning];
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]);
/* 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;
AudioStreamBasicDescription asbd = {0};
NSValue *asbdValue = [audio_format attributeForKey:QTFormatDescriptionAudioStreamBasicDescriptionAttribute];
[asbdValue getValue:&asbd];
msg_Dbg(p_demux, "New audio es %d channels %dHz",
audiofmt.audio.i_channels, audiofmt.audio.i_rate);
char formatIDString[5];
UInt32 formatID = CFSwapInt32HostToBig (asbd.mFormatID);
bcopy (&formatID, formatIDString, 4);
formatIDString[4] = '\0';
p_sys->p_es_audio = es_out_Add(p_demux->out, &audiofmt);
/* 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
}
[audioInput release];
[pool release];
if([audioformat_array count])
audio_format = [audioformat_array objectAtIndex:0];
else
goto error;
msg_Dbg(p_demux, "QTSound: We have an audio device ready!");
/* Now we can init */
audiocodec = VLC_CODEC_FL32;
es_format_Init(&audiofmt, AUDIO_ES, audiocodec);
return VLC_SUCCESS;
error:
[audioInput release];
[pool release];
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;
/*
* 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 / 8);
audiofmt.i_bitrate = audiofmt.audio.i_channels * audiofmt.audio.i_rate * audiofmt.audio.i_bitspersample;
free(p_sys);
p_sys->session = [[QTCaptureSession alloc] init];
return VLC_EGENERIC;
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 (%ld)", [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 (%ld)", [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];
msg_Dbg(p_demux, "QTSound: We have an audio device ready!");
return VLC_SUCCESS;
error:
[audioInput release];
free(p_sys);
return VLC_EGENERIC;
}
}
/*****************************************************************************
......@@ -475,17 +477,16 @@ error:
*****************************************************************************/
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];
@autoreleasepool {
demux_t *p_demux = (demux_t*)p_this;
demux_sys_t *p_sys = p_demux->p_sys;
free(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];
[pool release];
free(p_sys);
}
}
/*****************************************************************************
......@@ -495,25 +496,24 @@ static int Demux(demux_t *p_demux)
{
demux_sys_t *p_sys = p_demux->p_sys;
block_t *p_blocka = nil;
NSAutoreleasePool *pool;
@autoreleasepool {
@synchronized (p_sys->audiooutput) {
if ([p_sys->audiooutput checkCurrentAudioBuffer]) {
unsigned i_buffer_size = [p_sys->audiooutput getCurrentTotalDataSize];
p_blocka = block_Alloc(i_buffer_size);
@synchronized (p_sys->audiooutput) {
if ([p_sys->audiooutput checkCurrentAudioBuffer]) {
unsigned i_buffer_size = [p_sys->audiooutput getCurrentTotalDataSize];
p_blocka = block_Alloc(i_buffer_size);
if(!p_blocka) {
msg_Err(p_demux, "cannot get audio block");
return 0;
}
if(!p_blocka) {
msg_Err(p_demux, "cannot get audio block");
return 0;
memcpy(p_blocka->p_buffer, [p_sys->audiooutput getCurrentAudioBufferData], i_buffer_size);
p_blocka->i_nb_samples = [p_sys->audiooutput getNumberOfSamples];
p_blocka->i_pts = [p_sys->audiooutput getCurrentPts];
[p_sys->audiooutput freeAudioMem];
}
memcpy(p_blocka->p_buffer, [p_sys->audiooutput getCurrentAudioBufferData], i_buffer_size);
p_blocka->i_nb_samples = [p_sys->audiooutput getNumberOfSamples];
p_blocka->i_pts = [p_sys->audiooutput getCurrentPts];
[p_sys->audiooutput freeAudioMem];
}
}
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment