summaryrefslogtreecommitdiff
path: root/libavcodec/dss_sp.c
diff options
context:
space:
mode:
Diffstat (limited to 'libavcodec/dss_sp.c')
-rw-r--r--libavcodec/dss_sp.c53
1 files changed, 29 insertions, 24 deletions
diff --git a/libavcodec/dss_sp.c b/libavcodec/dss_sp.c
index 4fe784c055..14025fcdde 100644
--- a/libavcodec/dss_sp.c
+++ b/libavcodec/dss_sp.c
@@ -2,20 +2,20 @@
* Digital Speech Standard - Standard Play mode (DSS SP) audio decoder.
* Copyright (C) 2014 Oleksij Rempel <linux@rempel-privat.de>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
@@ -25,7 +25,7 @@
#include "libavutil/opt.h"
#include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "internal.h"
#define SUBFRAMES 4
@@ -33,7 +33,7 @@
#define DSS_SP_FRAME_SIZE 42
#define DSS_SP_SAMPLE_COUNT (66 * SUBFRAMES)
-#define DSS_SP_FORMULA(a, b, c) ((((a) << 15) + (b) * (c)) + 0x4000) >> 15
+#define DSS_SP_FORMULA(a, b, c) ((int)((((a) * (1 << 15)) + (b) * (unsigned)(c)) + 0x4000) >> 15)
typedef struct DssSpSubframe {
int16_t gain;
@@ -50,6 +50,7 @@ typedef struct DssSpFrame {
} DssSpFrame;
typedef struct DssSpContext {
+ AVCodecContext *avctx;
int32_t excitation[288 + 6];
int32_t history[187];
DssSpFrame fparam;
@@ -296,13 +297,14 @@ static av_cold int dss_sp_decode_init(AVCodecContext *avctx)
memset(p->history, 0, sizeof(p->history));
p->pulse_dec_mode = 1;
+ p->avctx = avctx;
return 0;
}
static void dss_sp_unpack_coeffs(DssSpContext *p, const uint8_t *src)
{
- BitstreamContext bc;
+ GetBitContext gb;
DssSpFrame *fparam = &p->fparam;
int i;
int subframe_idx;
@@ -315,24 +317,24 @@ static void dss_sp_unpack_coeffs(DssSpContext *p, const uint8_t *src)
p->bits[i + 1] = src[i];
}
- bitstream_init8(&bc, p->bits, DSS_SP_FRAME_SIZE);
+ init_get_bits(&gb, p->bits, DSS_SP_FRAME_SIZE * 8);
for (i = 0; i < 2; i++)
- fparam->filter_idx[i] = bitstream_read(&bc, 5);
+ fparam->filter_idx[i] = get_bits(&gb, 5);
for (; i < 8; i++)
- fparam->filter_idx[i] = bitstream_read(&bc, 4);
+ fparam->filter_idx[i] = get_bits(&gb, 4);
for (; i < 14; i++)
- fparam->filter_idx[i] = bitstream_read(&bc, 3);
+ fparam->filter_idx[i] = get_bits(&gb, 3);
for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) {
- fparam->sf_adaptive_gain[subframe_idx] = bitstream_read(&bc, 5);
+ fparam->sf_adaptive_gain[subframe_idx] = get_bits(&gb, 5);
- fparam->sf[subframe_idx].combined_pulse_pos = bitstream_read(&bc, 31);
+ fparam->sf[subframe_idx].combined_pulse_pos = get_bits_long(&gb, 31);
- fparam->sf[subframe_idx].gain = bitstream_read(&bc, 6);
+ fparam->sf[subframe_idx].gain = get_bits(&gb, 6);
for (i = 0; i < 7; i++)
- fparam->sf[subframe_idx].pulse_val[i] = bitstream_read(&bc, 3);
+ fparam->sf[subframe_idx].pulse_val[i] = get_bits(&gb, 3);
}
for (subframe_idx = 0; subframe_idx < 4; subframe_idx++) {
@@ -378,7 +380,7 @@ static void dss_sp_unpack_coeffs(DssSpContext *p, const uint8_t *src)
if (C72_binomials[index] <= combined_pulse_pos) {
combined_pulse_pos -= C72_binomials[index];
- fparam->sf[subframe_idx].pulse_pos[(index ^ 7) - 1] = i;
+ fparam->sf[subframe_idx].pulse_pos[6 - index] = i;
if (!index)
break;
@@ -394,16 +396,21 @@ static void dss_sp_unpack_coeffs(DssSpContext *p, const uint8_t *src)
}
}
- combined_pitch = bitstream_read(&bc, 24);
+ combined_pitch = get_bits(&gb, 24);
fparam->pitch_lag[0] = (combined_pitch % 151) + 36;
combined_pitch /= 151;
- for (i = 1; i < SUBFRAMES; i++) {
+ for (i = 1; i < SUBFRAMES - 1; i++) {
fparam->pitch_lag[i] = combined_pitch % 48;
combined_pitch /= 48;
}
+ if (combined_pitch > 47) {
+ av_log (p->avctx, AV_LOG_WARNING, "combined_pitch was too large\n");
+ combined_pitch = 0;
+ }
+ fparam->pitch_lag[i] = combined_pitch;
pitch_lag = fparam->pitch_lag[0];
for (i = 1; i < SUBFRAMES; i++) {
@@ -492,7 +499,7 @@ static void dss_sp_scale_vector(int32_t *vec, int bits, int size)
vec[i] = vec[i] >> -bits;
else
for (i = 0; i < size; i++)
- vec[i] = vec[i] << bits;
+ vec[i] = vec[i] * (1 << bits);
}
static void dss_sp_update_buf(int32_t *hist, int32_t *vector)
@@ -517,12 +524,12 @@ static void dss_sp_shift_sq_sub(const int32_t *filter_buf,
tmp = dst[a] * filter_buf[0];
for (i = 14; i > 0; i--)
- tmp -= error_buf[i] * filter_buf[i];
+ tmp -= error_buf[i] * (unsigned)filter_buf[i];
for (i = 14; i > 0; i--)
error_buf[i] = error_buf[i - 1];
- tmp = (tmp + 4096) >> 13;
+ tmp = (int)(tmp + 4096U) >> 13;
error_buf[1] = tmp;
@@ -754,10 +761,8 @@ static int dss_sp_decode_frame(AVCodecContext *avctx, void *data,
}
frame->nb_samples = DSS_SP_SAMPLE_COUNT;
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed.\n");
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
- }
out = (int16_t *)frame->data[0];
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-14 20:02:21 +0000