Add service locks for parallelism.

[mlt] / src / modules / avformat / filter_avresample.c

diff --git a/src/modules/avformat/filter_avresample.c b/src/modules/avformat/filter_avresample.c
index 7525004ba06e166f79e5ed2f010f3fd035936e10..f78dff87814a642da4cf37ce07ecbe6ce66ab174 100644 (file)
--- a/src/modules/avformat/filter_avresample.c
+++ b/src/modules/avformat/filter_avresample.c
@@ -18,8 +18,7 @@
  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  */
 
-#include "filter_avresample.h"
-
+#include <framework/mlt_filter.h>
 #include <framework/mlt_frame.h>
 
 #include <stdio.h>
@@ -27,12 +26,12 @@
 #include <string.h>
 
 // ffmpeg Header files
-#include <avformat.h>
+#include <libavformat/avformat.h>
 
 /** Get the audio.
 */
 
-static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
+static int resample_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
 {
        // Get the properties of the frame
        mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
@@ -43,6 +42,8 @@ static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
        // Get the filter properties
        mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
 
+       mlt_service_lock( MLT_FILTER_SERVICE( filter ) );
+
        // Get the resample information
        int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
        int16_t *sample_buffer = mlt_properties_get_data( filter_properties, "buffer", NULL );
@@ -61,6 +62,7 @@ static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
                output_rate = *frequency;
 
        // Get the producer's audio
+       *format = mlt_audio_s16;
        mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
 
        // Duplicate channels as necessary
@@ -75,7 +77,7 @@ static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
                {
                        for ( j = 0; j < *channels; j++ )
                        {
-                               new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
+                               new_buffer[ ( i * *channels ) + j ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + k ];
                                k = ( k + 1 ) % channels_avail;
                        }
                }
@@ -94,8 +96,8 @@ static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
                // Drop all but the first *channels
                for ( i = 0; i < *samples; i++ )
                {
-                       new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
-                       new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
+                       new_buffer[ ( i * *channels ) + 0 ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + 2 ];
+                       new_buffer[ ( i * *channels ) + 1 ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + 3 ];
                }
 
                // Update the audio buffer now - destroys the old
@@ -114,7 +116,12 @@ static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
                if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
                {
                        // Create the resampler
+#if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0))
+                       resample = av_audio_resample_init( *channels, *channels, output_rate, *frequency,
+                               SAMPLE_FMT_S16, SAMPLE_FMT_S16, 16, 10, 0, 0.8 );
+#else
                        resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
+#endif
 
                        // And store it on properties
                        mlt_properties_set_data( filter_properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
@@ -123,23 +130,30 @@ static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
                        mlt_properties_set_int( filter_properties, "last_frequency", *frequency );
                }
 
+               mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
+
                // Resample the audio
                used = audio_resample( resample, sample_buffer, *buffer, *samples );
+               int size = used * *channels * sizeof( int16_t );
 
                // Resize if necessary
                if ( used > *samples )
                {
-                       *buffer = mlt_pool_realloc( *buffer, *samples * *channels * sizeof( int16_t ) );
-                       mlt_properties_set_data( properties, "audio", *buffer, *channels * used * sizeof( int16_t ), mlt_pool_release, NULL );
+                       *buffer = mlt_pool_realloc( *buffer, size );
+                       mlt_frame_set_audio( frame, *buffer, *format, size, mlt_pool_release );
                }
 
                // Copy samples
-               memcpy( *buffer, sample_buffer, *channels * used * sizeof( int16_t ) );
+               memcpy( *buffer, sample_buffer, size );
 
                // Update output variables
                *samples = used;
                *frequency = output_rate;
        }
+       else
+       {
+               mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
+       }
 
        return 0;
 }