]> git.sesse.net Git - mlt/commitdiff
producer_avformat.c: bugfix audio sync with some codecs and revert unnecessary precau...
authorddennedy <ddennedy@d19143bc-622f-0410-bfdd-b5b2a6649095>
Tue, 26 Jun 2007 06:57:09 +0000 (06:57 +0000)
committerddennedy <ddennedy@d19143bc-622f-0410-bfdd-b5b2a6649095>
Tue, 26 Jun 2007 06:57:09 +0000 (06:57 +0000)
git-svn-id: https://mlt.svn.sourceforge.net/svnroot/mlt/trunk/mlt@991 d19143bc-622f-0410-bfdd-b5b2a6649095

src/modules/avformat/producer_avformat.c

index 0648bde494984209c8baa8ff1afdf86d4d31221c..bc7971243fb7e041062502a86327e54f9b463ac4 100644 (file)
@@ -617,17 +617,8 @@ static int producer_get_image( mlt_frame frame, uint8_t **buffer, mlt_image_form
 
                                // Decode the image
                                if ( must_decode || int_position >= req_position )
-                               {
-#ifdef FF_INPUT_BUFFER_PADDING_SIZE
-                                       uint8_t *inbuf = mlt_pool_alloc( pkt.size + FF_INPUT_BUFFER_PADDING_SIZE );
-                                       memcpy( inbuf, pkt.data, pkt.size );
-                                       inbuf[ pkt.size ] = 0;
-                                       ret = avcodec_decode_video( codec_context, av_frame, &got_picture, inbuf, pkt.size );
-                                       mlt_pool_release( inbuf );
-#else
                                        ret = avcodec_decode_video( codec_context, av_frame, &got_picture, pkt.data, pkt.size );
-#endif
-                               }
+
                                if ( got_picture )
                                {
                                        // Handle ignore
@@ -888,7 +879,7 @@ static int producer_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
        {
                int ret = 0;
                int got_audio = 0;
-               int16_t *temp = mlt_pool_alloc( sizeof( int16_t ) * AVCODEC_MAX_AUDIO_FRAME_SIZE );
+               int16_t *temp = av_malloc( sizeof( int16_t ) * AVCODEC_MAX_AUDIO_FRAME_SIZE );
 
                av_init_packet( &pkt );
 
@@ -904,24 +895,20 @@ static int producer_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
                        // Read a packet
                        ret = av_read_frame( context, &pkt );
 
-               int len = pkt.size;
-               uint8_t *ptr = pkt.data;
-                       int data_size = sizeof( int16_t ) * AVCODEC_MAX_AUDIO_FRAME_SIZE;
+                       int len = pkt.size;
+                       uint8_t *ptr = pkt.data;
 
                        // We only deal with audio from the selected audio_index
                        while ( ptr != NULL && ret >= 0 && pkt.stream_index == index && len > 0 )
                        {
+                               int data_size = sizeof( int16_t ) * AVCODEC_MAX_AUDIO_FRAME_SIZE;
+
                                // Decode the audio
 #if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(29<<8)+0))
-                               uint8_t *inbuf = mlt_pool_alloc( len + FF_INPUT_BUFFER_PADDING_SIZE );
-                               memcpy( inbuf, ptr, len );
-                               inbuf[ len ] = 0;
-                               ret = avcodec_decode_audio2( codec_context, temp, &data_size, inbuf, len );
-                               mlt_pool_release( inbuf );
+                               ret = avcodec_decode_audio2( codec_context, temp, &data_size, ptr, len );
 #else
                                ret = avcodec_decode_audio( codec_context, temp, &data_size, ptr, len );
 #endif
-
                                if ( ret < 0 )
                                {
                                        ret = 0;
@@ -981,7 +968,7 @@ static int producer_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
                mlt_properties_set_int( properties, "_audio_used", audio_used );
 
                // Release the temporary audio
-               mlt_pool_release( temp );
+               av_free( temp );
        }
        else
        {