]> git.sesse.net Git - ffmpeg/blobdiff - libavcodec/atrac3.c
rtmp: Factorize the code by adding handle_invoke_status
[ffmpeg] / libavcodec / atrac3.c
index bdd03402da0873b78b344b916268e50e5ac66c6d..a09007c20f45957bfbae4db58b8d9d6af5d2fce7 100644 (file)
@@ -36,9 +36,9 @@
 #include <stddef.h>
 #include <stdio.h>
 
+#include "libavutil/float_dsp.h"
 #include "avcodec.h"
 #include "get_bits.h"
-#include "dsputil.h"
 #include "bytestream.h"
 #include "fft.h"
 #include "fmtconvert.h"
@@ -125,13 +125,13 @@ typedef struct {
 
     FFTContext          mdct_ctx;
     FmtConvertContext   fmt_conv;
+    AVFloatDSPContext   fdsp;
 } ATRAC3Context;
 
 static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
 static VLC              spectral_coeff_tab[7];
 static float            gain_tab1[16];
 static float            gain_tab2[31];
-static DSPContext       dsp;
 
 
 /**
@@ -164,7 +164,7 @@ static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
     q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);
 
     /* Perform windowing on the output. */
-    dsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE);
+    q->fdsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE);
 
 }
 
@@ -402,6 +402,8 @@ static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent
 
             for (k=0; k<coded_components; k++) {
                 sfIndx = get_bits(gb,6);
+                if (component_count >= 64)
+                    return AVERROR_INVALIDDATA;
                 pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
                 max_coded_values = SAMPLES_PER_FRAME - pComponent[component_count].pos;
                 coded_values = coded_values_per_component + 1;
@@ -708,9 +710,10 @@ static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_
             memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
 
         /* gain compensation and overlapping */
-        gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
-                                    &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
-                                    &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
+        gainCompensateAndOverlap(pSnd->IMDCT_buf, &pSnd->prevFrame[band * 256],
+                                 &pOut[band * 256],
+                                 &pSnd->gainBlock[1 - pSnd->gcBlkSwitch].gBlock[band],
+                                 &pSnd->gainBlock[    pSnd->gcBlkSwitch].gBlock[band]);
     }
 
     /* Swap the gain control buffers for the next frame. */
@@ -741,7 +744,7 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
 
         result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, out_samples[0], 0, JOINT_STEREO);
         if (result != 0)
-            return (result);
+            return result;
 
         /* Framedata of the su2 in the joint-stereo mode is encoded in
          * reverse byte order so we need to swap it first. */
@@ -782,7 +785,7 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
         /* Decode Sound Unit 2. */
         result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], out_samples[1], 1, JOINT_STEREO);
         if (result != 0)
-            return (result);
+            return result;
 
         /* Reconstruct the channel coefficients. */
         reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
@@ -795,11 +798,13 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
         for (i=0 ; i<q->channels ; i++) {
 
             /* Set the bitstream reader at the start of a channel sound unit. */
-            init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
+            init_get_bits(&q->gb,
+                          databuf + i * q->bytes_per_frame / q->channels,
+                          q->bits_per_frame / q->channels);
 
             result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], out_samples[i], i, q->codingMode);
             if (result != 0)
-                return (result);
+                return result;
         }
     }
 
@@ -809,9 +814,9 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
         p2= p1+256;
         p3= p2+256;
         p4= p3+256;
-        atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
-        atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
-        atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
+        ff_atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
+        ff_atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
+        ff_atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
     }
 
     return 0;
@@ -1011,7 +1016,7 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
         return ret;
     }
 
-    atrac_generate_tables();
+    ff_atrac_generate_tables();
 
     /* Generate gain tables. */
     for (i=0 ; i<16 ; i++)
@@ -1034,7 +1039,7 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
         q->matrix_coeff_index_next[i] = 3;
     }
 
-    dsputil_init(&dsp, avctx);
+    avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
     ff_fmt_convert_init(&q->fmt_conv, avctx);
 
     q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
@@ -1061,13 +1066,13 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
 
 AVCodec ff_atrac3_decoder =
 {
-    .name = "atrac3",
-    .type = AVMEDIA_TYPE_AUDIO,
-    .id CODEC_ID_ATRAC3,
+    .name           = "atrac3",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_ATRAC3,
     .priv_data_size = sizeof(ATRAC3Context),
-    .init = atrac3_decode_init,
-    .close = atrac3_decode_close,
-    .decode = atrac3_decode_frame,
-    .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
-    .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
+    .init           = atrac3_decode_init,
+    .close          = atrac3_decode_close,
+    .decode         = atrac3_decode_frame,
+    .capabilities   = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
 };