Commit 152693ff authored by Michael Feurstein's avatar Michael Feurstein Committed by Jean-Baptiste Kempf

qtsound: fix broken qtsound module

Signed-off-by: default avatarJean-Baptiste Kempf <jb@videolan.org>
parent afc252c4
...@@ -165,12 +165,8 @@ vlc_module_end () ...@@ -165,12 +165,8 @@ vlc_module_end ()
b2Ptr = (const float *) tempAudioBufferList->mBuffers[1].mData; b2Ptr = (const float *) tempAudioBufferList->mBuffers[1].mData;
i < numberOfSamples; i++) i < numberOfSamples; i++)
{ {
*uPtr = *b1Ptr; *uPtr++ = *b1Ptr++;
uPtr ++; *uPtr++ = *b2Ptr++;
b1Ptr ++;
*uPtr = *b2Ptr;
uPtr ++;
b2Ptr ++;
} }
if (currentAudioBuffer == nil) if (currentAudioBuffer == nil)
...@@ -449,15 +445,6 @@ static int Open( vlc_object_t *p_this ) ...@@ -449,15 +445,6 @@ static int Open( vlc_object_t *p_this )
* before downmixing, the buffer is derived. * before downmixing, the buffer is derived.
*/ */
audiofmt.audio.i_original_channels = AOUT_CHAN_RIGHT | AOUT_CHAN_LEFT; 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 * Please note that it may be completely arbitrary - buffers are not
* obliged to contain a integral number of so-called "frames". It's * obliged to contain a integral number of so-called "frames". It's
...@@ -466,9 +453,9 @@ static int Open( vlc_object_t *p_this ) ...@@ -466,9 +453,9 @@ static int Open( vlc_object_t *p_this )
*/ */
audiofmt.audio.i_bitspersample = 32; audiofmt.audio.i_bitspersample = 32;
audiofmt.audio.i_channels = 2; audiofmt.audio.i_channels = 2;
audiofmt.audio.i_blockalign = audiofmt.audio.i_channels * audiofmt.audio.i_bitspersample / 16; 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; 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->i_audio_max_buffer_size = audiofmt.i_bitrate;
p_sys->session = [[QTCaptureSession alloc] init]; p_sys->session = [[QTCaptureSession alloc] init];
...@@ -554,10 +541,10 @@ static int Demux( demux_t *p_demux ) ...@@ -554,10 +541,10 @@ static int Demux( demux_t *p_demux )
{ {
if ( [p_sys->audiooutput checkCurrentAudioBuffer] ) if ( [p_sys->audiooutput checkCurrentAudioBuffer] )
{ {
p_blocka->i_pts = [p_sys->audiooutput getCurrentPts]; p_blocka->i_buffer = p_blocka->i_size = [p_sys->audiooutput getCurrentTotalDataSize];
p_blocka->p_buffer = [p_sys->audiooutput getCurrentAudioBufferData]; p_blocka->p_buffer = p_blocka->p_start = [p_sys->audiooutput getCurrentAudioBufferData];
p_blocka->i_nb_samples = [p_sys->audiooutput getNumberOfSamples]; p_blocka->i_nb_samples = [p_sys->audiooutput getNumberOfSamples];
p_blocka->i_buffer = [p_sys->audiooutput getCurrentTotalDataSize]; p_blocka->i_pts = [p_sys->audiooutput getCurrentPts];
} }
} }
...@@ -577,13 +564,6 @@ static int Demux( demux_t *p_demux ) ...@@ -577,13 +564,6 @@ static int Demux( demux_t *p_demux )
@synchronized (p_sys->audiooutput) @synchronized (p_sys->audiooutput)
{ {
/*
* Free Memory
*
* Wait before freeing memory, so we don't get no crackling sound
* crackling sound artefacts start at 100 ms and below
*/
msleep( 200 );
[p_sys->audiooutput freeAudioMem]; [p_sys->audiooutput freeAudioMem];
} }
......
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