[FFmpeg-cvslog] r9031 - in trunk/libavcodec: mpegaudiodec.c qdm2.c

[reimar]

Author: reimar
Date: Tue May 15 22:47:55 2007
New Revision: 9031

Log:
Use DECLARE_ALIGNED to ease porting


Modified:
   trunk/libavcodec/mpegaudiodec.c
   trunk/libavcodec/qdm2.c

Modified: trunk/libavcodec/mpegaudiodec.c
==============================================================================
--- trunk/libavcodec/mpegaudiodec.c	(original)
+++ trunk/libavcodec/mpegaudiodec.c	Tue May 15 22:47:55 2007
@@ -79,9 +79,9 @@ typedef struct MPADecodeContext {
     int mode;
     int mode_ext;
     int lsf;
-    MPA_INT synth_buf[MPA_MAX_CHANNELS][512 * 2] __attribute__((aligned(16)));
+    DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512 * 2]);
     int synth_buf_offset[MPA_MAX_CHANNELS];
-    int32_t sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT] __attribute__((aligned(16)));
+    DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]);
     int32_t mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
 #ifdef DEBUG
     int frame_count;
@@ -170,7 +170,7 @@ static const int32_t scale_factor_mult2[
     SCALE_GEN(4.0 / 9.0), /* 9 steps */
 };
 
-static MPA_INT window[512] __attribute__((aligned(16)));
+static DECLARE_ALIGNED_16(MPA_INT, window[512]);
 
 /* layer 1 unscaling */
 /* n = number of bits of the mantissa minus 1 */

Modified: trunk/libavcodec/qdm2.c
==============================================================================
--- trunk/libavcodec/qdm2.c	(original)
+++ trunk/libavcodec/qdm2.c	Tue May 15 22:47:55 2007
@@ -129,7 +129,7 @@ typedef struct {
 } QDM2Complex;
 
 typedef struct {
-    QDM2Complex complex[256 + 1] __attribute__((aligned(16)));
+    DECLARE_ALIGNED_16(QDM2Complex, complex[256 + 1]);
     float       samples_im[MPA_MAX_CHANNELS][256];
     float       samples_re[MPA_MAX_CHANNELS][256];
 } QDM2FFT;
@@ -182,9 +182,9 @@ typedef struct {
     float output_buffer[1024];
 
     /// Synthesis filter
-    MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
+    DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
     int synth_buf_offset[MPA_MAX_CHANNELS];
-    int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16)));
+    DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]);
 
     /// Mixed temporary data used in decoding
     float tone_level[MPA_MAX_CHANNELS][30][64];
@@ -229,7 +229,7 @@ static uint8_t random_dequant_index[256]
 static uint8_t random_dequant_type24[128][3];
 static float noise_samples[128];
 
-static MPA_INT mpa_window[512] __attribute__((aligned(16)));
+static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
 
 
 static void softclip_table_init(void) {