* Copyright (c) 2006 Benjamin Larsson
* Copyright (c) 2006 Konstantin Shishkov
*
- * 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
*/
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/internal.h"
+#include "libavutil/libm.h"
#include "avcodec.h"
#include "bswapdsp.h"
#include "get_bits.h"
GetBitContext gb;
BswapDSPContext bdsp;
- AVFloatDSPContext fdsp;
+ AVFloatDSPContext *fdsp;
FFTContext fft;
DECLARE_ALIGNED(32, FFTComplex, samples)[COEFFS / 2];
float *out_samples;
if (i > 0) {
tb = bark - prev_bark;
- q->weights1[i - 1] = pow(10.0, -1.0 * tb);
- q->weights2[i - 1] = pow(10.0, -2.7 * tb);
+ q->weights1[i - 1] = ff_exp10(-1.0 * tb);
+ q->weights2[i - 1] = ff_exp10(-2.7 * tb);
}
prev_bark = bark;
IMCContext *q = avctx->priv_data;
double r1, r2;
+ if (avctx->codec_id == AV_CODEC_ID_IAC && avctx->sample_rate > 96000) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Strange sample rate of %i, file likely corrupt or "
+ "needing a new table derivation method.\n",
+ avctx->sample_rate);
+ return AVERROR_PATCHWELCOME;
+ }
+
if (avctx->codec_id == AV_CODEC_ID_IMC)
avctx->channels = 1;
return ret;
}
ff_bswapdsp_init(&q->bdsp);
- avpriv_float_dsp_init(&q->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
+ q->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+ if (!q->fdsp) {
+ ff_fft_end(&q->fft);
+
+ return AVERROR(ENOMEM);
+ }
+
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO
: AV_CH_LAYOUT_STEREO;
float tmp, tmp2;
// maybe some frequency division thingy
- flcoeffs1[0] = 20000.0 / pow (2, levlCoeffBuf[0] * 0.18945); // 0.18945 = log2(10) * 0.05703125
+ flcoeffs1[0] = 20000.0 / exp2 (levlCoeffBuf[0] * 0.18945); // 0.18945 = log2(10) * 0.05703125
flcoeffs2[0] = log2f(flcoeffs1[0]);
tmp = flcoeffs1[0];
tmp2 = flcoeffs2[0];
for (i = 0; i < BANDS; i++)
highest = FFMAX(highest, chctx->flcoeffs1[i]);
- for (i = 0; i < BANDS - 1; i++)
+ for (i = 0; i < BANDS - 1; i++) {
+ if (chctx->flcoeffs5[i] <= 0) {
+ av_log(NULL, AV_LOG_ERROR, "flcoeffs5 %f invalid\n", chctx->flcoeffs5[i]);
+ return AVERROR_INVALIDDATA;
+ }
chctx->flcoeffs4[i] = chctx->flcoeffs3[i] - log2f(chctx->flcoeffs5[i]);
+ }
chctx->flcoeffs4[BANDS - 1] = limit;
highest = highest * 0.25;
}
-static int imc_get_coeffs(IMCContext *q, IMCChannel *chctx)
+static void imc_get_coeffs(AVCodecContext *avctx,
+ IMCContext *q, IMCChannel *chctx)
{
int i, j, cw_len, cw;
cw_len = chctx->CWlengthT[j];
cw = 0;
- if (get_bits_count(&q->gb) + cw_len > 512) {
- ff_dlog(NULL, "Band %i coeff %i cw_len %i\n", i, j, cw_len);
- return AVERROR_INVALIDDATA;
+ if (cw_len && (!chctx->bandFlagsBuf[i] || !chctx->skipFlags[j])) {
+ if (get_bits_count(&q->gb) + cw_len > 512) {
+ av_log(avctx, AV_LOG_WARNING,
+ "Potential problem on band %i, coefficient %i"
+ ": cw_len=%i\n", i, j, cw_len);
+ } else
+ cw = get_bits(&q->gb, cw_len);
}
- if (cw_len && (!chctx->bandFlagsBuf[i] || !chctx->skipFlags[j]))
- cw = get_bits(&q->gb, cw_len);
-
chctx->codewords[j] = cw;
}
}
}
- return 0;
}
static void imc_refine_bit_allocation(IMCContext *q, IMCChannel *chctx)
imc_decode_level_coefficients2(q, chctx->levlCoeffBuf, chctx->old_floor,
chctx->flcoeffs1, chctx->flcoeffs2);
+ for(i=0; i<BANDS; i++) {
+ if(chctx->flcoeffs1[i] > INT_MAX) {
+ av_log(avctx, AV_LOG_ERROR, "scalefactor out of range\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
memcpy(chctx->old_floor, chctx->flcoeffs1, 32 * sizeof(float));
counter = 0;
memset(chctx->codewords, 0, sizeof(chctx->codewords));
- if (imc_get_coeffs(q, chctx) < 0) {
- av_log(avctx, AV_LOG_ERROR, "Read coefficients failed\n");
- chctx->decoder_reset = 1;
- return AVERROR_INVALIDDATA;
- }
+ imc_get_coeffs(avctx, q, chctx);
if (inverse_quant_coeff(q, chctx, stream_format_code) < 0) {
av_log(avctx, AV_LOG_ERROR, "Inverse quantization of coefficients failed\n");
/* get output buffer */
frame->nb_samples = COEFFS;
- 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;
- }
for (i = 0; i < avctx->channels; i++) {
q->out_samples = (float *)frame->extended_data[i];
}
if (avctx->channels == 2) {
- q->fdsp.butterflies_float((float *)frame->extended_data[0],
+ q->fdsp->butterflies_float((float *)frame->extended_data[0],
(float *)frame->extended_data[1], COEFFS);
}
return IMC_BLOCK_SIZE * avctx->channels;
}
-
static av_cold int imc_decode_close(AVCodecContext * avctx)
{
IMCContext *q = avctx->priv_data;
ff_fft_end(&q->fft);
+ av_freep(&q->fdsp);
return 0;
}
+static av_cold void flush(AVCodecContext *avctx)
+{
+ IMCContext *q = avctx->priv_data;
+
+ q->chctx[0].decoder_reset =
+ q->chctx[1].decoder_reset = 1;
+}
+#if CONFIG_IMC_DECODER
AVCodec ff_imc_decoder = {
.name = "imc",
.long_name = NULL_IF_CONFIG_SMALL("IMC (Intel Music Coder)"),
.init = imc_decode_init,
.close = imc_decode_close,
.decode = imc_decode_frame,
+ .flush = flush,
.capabilities = AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
-
+#endif
+#if CONFIG_IAC_DECODER
AVCodec ff_iac_decoder = {
.name = "iac",
.long_name = NULL_IF_CONFIG_SMALL("IAC (Indeo Audio Coder)"),
.init = imc_decode_init,
.close = imc_decode_close,
.decode = imc_decode_frame,
+ .flush = flush,
.capabilities = AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
+#endif
projects / ffmpeg / blobdiff