X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fcook.c;h=f487db6a975b41ceb08f91f2dc55b16dcef791a8;hb=1f1ad8ace040a08edc2646ff638ca42a8828779f;hp=90d78eb3f562d33188a5afeb3d91c584cc0bea5d;hpb=6d97484d72e33f7dde9493a9ead1a72e2f029605;p=ffmpeg

diff --git a/libavcodec/cook.c b/libavcodec/cook.c
index 90d78eb3f56..f487db6a975 100644
--- a/libavcodec/cook.c
+++ b/libavcodec/cook.c
@@ -44,9 +44,10 @@
 
 #include "libavutil/channel_layout.h"
 #include "libavutil/lfg.h"
+
+#include "audiodsp.h"
 #include "avcodec.h"
 #include "get_bits.h"
-#include "dsputil.h"
 #include "bytestream.h"
 #include "fft.h"
 #include "internal.h"
@@ -63,12 +64,12 @@
 #define SUBBAND_SIZE    20
 #define MAX_SUBPACKETS   5
 
-typedef struct {
+typedef struct cook_gains {
     int *now;
     int *previous;
 } cook_gains;
 
-typedef struct {
+typedef struct COOKSubpacket {
     int                 ch_idx;
     int                 size;
     int                 num_channels;
@@ -122,7 +123,7 @@ typedef struct cook {
     void (*saturate_output)(struct cook *q, float *out);
 
     AVCodecContext*     avctx;
-    DSPContext          dsp;
+    AudioDSPContext     adsp;
     GetBitContext       gb;
     /* stream data */
     int                 num_vectors;
@@ -139,7 +140,7 @@ typedef struct cook {
     VLC                 envelope_quant_index[13];
     VLC                 sqvh[7];          // scalar quantization
 
-    /* generatable tables and related variables */
+    /* generate tables and related variables */
     int                 gain_size_factor;
     float               gain_table[23];
 
@@ -865,8 +866,8 @@ static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
  */
 static void saturate_output_float(COOKContext *q, float *out)
 {
-    q->dsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
-                        -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
+    q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
+                         -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
 }
 
 
@@ -1012,14 +1013,14 @@ static int cook_decode_frame(AVCodecContext *avctx, void *data,
 static void dump_cook_context(COOKContext *q)
 {
     //int i=0;
-#define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
-    av_dlog(q->avctx, "COOKextradata\n");
-    av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
+#define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
+    ff_dlog(q->avctx, "COOKextradata\n");
+    ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
     if (q->subpacket[0].cookversion > STEREO) {
         PRINT("js_subband_start", q->subpacket[0].js_subband_start);
         PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
     }
-    av_dlog(q->avctx, "COOKContext\n");
+    ff_dlog(q->avctx, "COOKContext\n");
     PRINT("nb_channels", q->avctx->channels);
     PRINT("bit_rate", q->avctx->bit_rate);
     PRINT("sample_rate", q->avctx->sample_rate);
@@ -1050,7 +1051,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
     q->avctx = avctx;
 
     /* Take care of the codec specific extradata. */
-    if (extradata_size <= 0) {
+    if (extradata_size < 8) {
         av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
         return AVERROR_INVALIDDATA;
     }
@@ -1065,7 +1066,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
     /* Initialize RNG. */
     av_lfg_init(&q->random_state, 0);
 
-    ff_dsputil_init(&q->dsp, avctx);
+    ff_audiodsp_init(&q->adsp);
 
     while (edata_ptr < edata_ptr_end) {
         /* 8 for mono, 16 for stereo, ? for multichannel
@@ -1169,7 +1170,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
         /* Initialize variable relations */
         q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
 
-        /* Try to catch some obviously faulty streams, othervise it might be exploitable */
+        /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
         if (q->subpacket[s].total_subbands > 53) {
             avpriv_request_sample(avctx, "total_subbands > 53");
             return AVERROR_PATCHWELCOME;
@@ -1212,12 +1213,12 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
 
     /* Pad the databuffer with:
        DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
-       FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
+       AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
     q->decoded_bytes_buffer =
         av_mallocz(avctx->block_align
                    + DECODE_BYTES_PAD1(avctx->block_align)
-                   + FF_INPUT_BUFFER_PADDING_SIZE);
-    if (q->decoded_bytes_buffer == NULL)
+                   + AV_INPUT_BUFFER_PADDING_SIZE);
+    if (!q->decoded_bytes_buffer)
         return AVERROR(ENOMEM);
 
     /* Initialize transform. */
@@ -1233,7 +1234,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
         q->saturate_output = saturate_output_float;
     }
 
-    /* Try to catch some obviously faulty streams, othervise it might be exploitable */
+    /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
     if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
         q->samples_per_channel != 1024) {
         avpriv_request_sample(avctx, "samples_per_channel = %d",
@@ -1255,14 +1256,14 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
 
 AVCodec ff_cook_decoder = {
     .name           = "cook",
+    .long_name      = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_COOK,
     .priv_data_size = sizeof(COOKContext),
     .init           = cook_decode_init,
     .close          = cook_decode_close,
     .decode         = cook_decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
+    .capabilities   = AV_CODEC_CAP_DR1,
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
 };