Use "const" qualifier for pointers that point to input data of

audio encoders.
This is purely for clarity/documentation purposes.

Originally committed as revision 24481 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavcodec/ac3enc.c

@@ -1181,7 +1181,7 @@ static int AC3_encode_frame(AVCodecContext *avctx,
                             unsigned char *frame, int buf_size, void *data)
 {
     AC3EncodeContext *s = avctx->priv_data;
-    int16_t *samples = data;
+    const int16_t *samples = data;
     int i, j, k, v, ch;
     int16_t input_samples[N];
     int32_t mdct_coef[NB_BLOCKS][AC3_MAX_CHANNELS][N/2];
@@ -1197,7 +1197,7 @@ static int AC3_encode_frame(AVCodecContext *avctx,
         int ich = s->channel_map[ch];
         /* fixed mdct to the six sub blocks & exponent computation */
         for(i=0;i<NB_BLOCKS;i++) {
-            int16_t *sptr;
+            const int16_t *sptr;
             int sinc;
 
             /* compute input samples */

libavcodec/alacenc.c

@@ -75,12 +75,12 @@ typedef struct AlacEncodeContext {
 } AlacEncodeContext;
 
 
-static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
+static void init_sample_buffers(AlacEncodeContext *s, const int16_t *input_samples)
 {
     int ch, i;
 
     for(ch=0;ch<s->avctx->channels;ch++) {
-        int16_t *sptr = input_samples + ch;
+        const int16_t *sptr = input_samples + ch;
         for(i=0;i<s->avctx->frame_size;i++) {
             s->sample_buf[ch][i] = *sptr;
             sptr += s->avctx->channels;
@@ -482,7 +482,7 @@ verbatim:
 
     if((s->compression_level == 0) || verbatim_flag) {
         // Verbatim mode
-        int16_t *samples = data;
+        const int16_t *samples = data;
         write_frame_header(s, 1);
         for(i=0; i<avctx->frame_size*avctx->channels; i++) {
             put_sbits(pb, 16, *samples++);

libavcodec/flacenc.c

@@ -446,7 +446,7 @@ static void init_frame(FlacEncodeContext *s)
 /**
  * Copy channel-interleaved input samples into separate subframes
  */
-static void copy_samples(FlacEncodeContext *s, int16_t *samples)
+static void copy_samples(FlacEncodeContext *s, const int16_t *samples)
 {
     int i, j, ch;
     FlacFrame *frame;
@@ -1204,7 +1204,7 @@ static void output_frame_footer(FlacEncodeContext *s)
     flush_put_bits(&s->pb);
 }
 
-static void update_md5_sum(FlacEncodeContext *s, int16_t *samples)
+static void update_md5_sum(FlacEncodeContext *s, const int16_t *samples)
 {
 #if HAVE_BIGENDIAN
     int i;
@@ -1213,7 +1213,7 @@ static void update_md5_sum(FlacEncodeContext *s, int16_t *samples)
         av_md5_update(s->md5ctx, (uint8_t *)&smp, 2);
     }
 #else
-    av_md5_update(s->md5ctx, (uint8_t *)samples, s->frame.blocksize*s->channels*2);
+    av_md5_update(s->md5ctx, (const uint8_t *)samples, s->frame.blocksize*s->channels*2);
 #endif
 }
 
@@ -1222,7 +1222,7 @@ static int flac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
 {
     int ch;
     FlacEncodeContext *s;
-    int16_t *samples = data;
+    const int16_t *samples = data;
     int out_bytes;
     int reencoded=0;
 

libavcodec/g726.c

@@ -348,7 +348,7 @@ static int g726_encode_frame(AVCodecContext *avctx,
                             uint8_t *dst, int buf_size, void *data)
 {
     G726Context *c = avctx->priv_data;
-    short *samples = data;
+    const short *samples = data;
     PutBitContext pb;
 
     init_put_bits(&pb, dst, 1024*1024);

libavcodec/mpegaudioenc.c

@@ -306,7 +306,7 @@ static void idct32(int *out, int *tab)
 
 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
 
-static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
+static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
 {
     short *p, *q;
     int sum, offset, i, j;
@@ -752,7 +752,7 @@ static int MPA_encode_frame(AVCodecContext *avctx,
                             unsigned char *frame, int buf_size, void *data)
 {
     MpegAudioContext *s = avctx->priv_data;
-    short *samples = data;
+    const short *samples = data;
     short smr[MPA_MAX_CHANNELS][SBLIMIT];
     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
     int padding, i;

libavcodec/nellymoserenc.c

@@ -351,7 +351,7 @@ static void encode_block(NellyMoserEncodeContext *s, unsigned char *output, int
 static int encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size, void *data)
 {
     NellyMoserEncodeContext *s = avctx->priv_data;
-    int16_t *samples = data;
+    const int16_t *samples = data;
     int i;
 
     if (s->last_frame)

libavcodec/pcm.c

@@ -81,14 +81,14 @@ static int pcm_encode_frame(AVCodecContext *avctx,
                             unsigned char *frame, int buf_size, void *data)
 {
     int n, sample_size, v;
-    short *samples;
+    const short *samples;
     unsigned char *dst;
-    uint8_t *srcu8;
+    const uint8_t *srcu8;
-    int16_t *samples_int16_t;
+    const int16_t *samples_int16_t;
-    int32_t *samples_int32_t;
+    const int32_t *samples_int32_t;
-    int64_t *samples_int64_t;
+    const int64_t *samples_int64_t;
-    uint16_t *samples_uint16_t;
+    const uint16_t *samples_uint16_t;
-    uint32_t *samples_uint32_t;
+    const uint32_t *samples_uint32_t;
 
     sample_size = av_get_bits_per_sample(avctx->codec->id)/8;
     n = buf_size / sample_size;

libavcodec/roqaudioenc.c

@@ -108,7 +108,7 @@ static int roq_dpcm_encode_frame(AVCodecContext *avctx,
                 unsigned char *frame, int buf_size, void *data)
 {
     int i, samples, stereo, ch;
-    short *in;
+    const short *in;
     unsigned char *out;
 
     ROQDPCMContext *context = avctx->priv_data;

libavcodec/vorbis_enc.c

@@ -888,7 +888,7 @@ static void residue_encode(vorbis_enc_context *venc, vorbis_enc_residue *rc,
     }
 }
 
-static int apply_window_and_mdct(vorbis_enc_context *venc, signed short *audio,
+static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *audio,
                                  int samples)
 {
     int i, j, channel;
@@ -973,7 +973,7 @@ static int vorbis_encode_frame(AVCodecContext *avccontext,
                                int buf_size, void *data)
 {
     vorbis_enc_context *venc = avccontext->priv_data;
-    signed short *audio = data;
+    const signed short *audio = data;
     int samples = data ? avccontext->frame_size : 0;
     vorbis_enc_mode *mode;
     vorbis_enc_mapping *mapping;

libavcodec/wmaenc.c

@@ -74,7 +74,7 @@ static int encode_init(AVCodecContext * avctx){
 }
 
 
-static void apply_window_and_mdct(AVCodecContext * avctx, signed short * audio, int len) {
+static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * audio, int len) {
     WMACodecContext *s = avctx->priv_data;
     int window_index= s->frame_len_bits - s->block_len_bits;
     int i, j, channel;
@@ -328,7 +328,7 @@ static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
 static int encode_superframe(AVCodecContext *avctx,
                             unsigned char *buf, int buf_size, void *data){
     WMACodecContext *s = avctx->priv_data;
-    short *samples = data;
+    const short *samples = data;
     int i, total_gain;
 
     s->block_len_bits= s->frame_len_bits; //required by non variable block len