ac3enc: move ff_ac3_encode_frame() to ac3enc_template.c

[ffmpeg] / libavcodec / amrnbdec.c

diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c
index 40cd91d1103200226a5bdef0ff30508b4eb0849c..4e24e26d68e9f8c05f5571accd836a3da4096401 100644 (file)
--- a/libavcodec/amrnbdec.c
+++ b/libavcodec/amrnbdec.c
@@ -3,20 +3,20 @@
  * Copyright (c) 2006-2007 Robert Swain
  * Copyright (c) 2009 Colin McQuillan
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -52,6 +52,7 @@
 #include "acelp_vectors.h"
 #include "acelp_pitch_delay.h"
 #include "lsp.h"
+#include "amr.h"
 
 #include "amrnbdata.h"
 
@@ -153,7 +154,7 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx)
     AMRContext *p = avctx->priv_data;
     int i;
 
-    avctx->sample_fmt = SAMPLE_FMT_FLT;
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
 
     // p->excitation always points to the same position in p->excitation_buf
     p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
@@ -195,24 +196,9 @@ static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
     p->bad_frame_indicator = !get_bits1(&gb); // quality bit
     skip_bits(&gb, 2);                        // two padding bits
 
-    if (mode < MODE_DTX) {
-        uint16_t *data = (uint16_t *)&p->frame;
-        const uint8_t *order = amr_unpacking_bitmaps_per_mode[mode];
-        int field_size;
-
-        memset(&p->frame, 0, sizeof(AMRNBFrame));
-        buf++;
-        while ((field_size = *order++)) {
-            int field = 0;
-            int field_offset = *order++;
-            while (field_size--) {
-               int bit = *order++;
-               field <<= 1;
-               field |= buf[bit >> 3] >> (bit & 7) & 1;
-            }
-            data[field_offset] = field;
-        }
-    }
+    if (mode < MODE_DTX)
+        ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
+                           amr_unpacking_bitmaps_per_mode[mode]);
 
     return mode;
 }
@@ -221,20 +207,6 @@ static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
 /// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions
 /// @{
 
-/**
- * Convert an lsf vector into an lsp vector.
- *
- * @param lsf               input lsf vector
- * @param lsp               output lsp vector
- */
-static void lsf2lsp(const float *lsf, double *lsp)
-{
-    int i;
-
-    for (i = 0; i < LP_FILTER_ORDER; i++)
-        lsp[i] = cos(2.0 * M_PI * lsf[i]);
-}
-
 /**
  * Interpolate the LSF vector (used for fixed gain smoothing).
  * The interpolation is done over all four subframes even in MODE_12k2.
@@ -283,7 +255,7 @@ static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
     }
 
     if (update)
-        memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(float));
+        memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
 
     for (i = 0; i < LP_FILTER_ORDER; i++)
         lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
@@ -293,7 +265,7 @@ static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
     if (update)
         interpolate_lsf(p->lsf_q, lsf_q);
 
-    lsf2lsp(lsf_q, lsp);
+    ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
 }
 
 /**
@@ -357,7 +329,7 @@ static void lsf2lsp_3(AMRContext *p)
     interpolate_lsf(p->lsf_q, lsf_q);
     memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
 
-    lsf2lsp(lsf_q, p->lsp[3]);
+    ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
 
     // interpolate LSP vectors at subframes 1, 2 and 3
     for (i = 1; i <= 3; i++)
@@ -796,7 +768,7 @@ static int synthesis(AMRContext *p, float *lpc,
                      float fixed_gain, const float *fixed_vector,
                      float *samples, uint8_t overflow)
 {
-    int i, overflow_temp = 0;
+    int i;
     float excitation[AMR_SUBFRAME_SIZE];
 
     // if an overflow has been detected, the pitch vector is scaled down by a
@@ -831,12 +803,10 @@ static int synthesis(AMRContext *p, float *lpc,
     // detect overflow
     for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
         if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
-            overflow_temp = 1;
-            samples[i] = av_clipf(samples[i], -AMR_SAMPLE_BOUND,
-                                               AMR_SAMPLE_BOUND);
+            return 1;
         }
 
-    return overflow_temp;
+    return 0;
 }
 
 /// @}
@@ -1045,13 +1015,10 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
     }
 
     ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
-                                             highpass_poles, highpass_gain,
+                                             highpass_poles,
+                                             highpass_gain * AMR_SAMPLE_SCALE,
                                              p->high_pass_mem, AMR_BLOCK_SIZE);
 
-    for (i = 0; i < AMR_BLOCK_SIZE; i++)
-        buf_out[i] = av_clipf(buf_out[i] * AMR_SAMPLE_SCALE,
-                              -1.0, 32767.0 / 32768.0);
-
     /* Update averaged lsf vector (used for fixed gain smoothing).
      *
      * Note that lsf_avg should not incorporate the current frame's LSFs
@@ -1069,7 +1036,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
 }
 
 
-AVCodec amrnb_decoder = {
+AVCodec ff_amrnb_decoder = {
     .name           = "amrnb",
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = CODEC_ID_AMR_NB,
@@ -1077,5 +1044,5 @@ AVCodec amrnb_decoder = {
     .init           = amrnb_decode_init,
     .decode         = amrnb_decode_frame,
     .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
-    .sample_fmts    = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},
+    .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
 };