/*
* Bink Audio decoder
- * Copyright (c) 2007-2010 Peter Ross (pross@xvid.org)
+ * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
* Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
*
- * 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
*/
/**
- * @file libavcodec/binkaudio.c
+ * @file
* Bink Audio decoder
*
* Technical details here:
*/
#include "avcodec.h"
-#define ALT_BITSTREAM_READER_LE
+#define BITSTREAM_READER_LE
#include "get_bits.h"
#include "dsputil.h"
+#include "dct.h"
+#include "rdft.h"
+#include "fmtconvert.h"
+#include "libavutil/intfloat.h"
+
extern const uint16_t ff_wma_critical_freqs[25];
+static float quant_table[96];
+
#define MAX_CHANNELS 2
#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
typedef struct {
- AVCodecContext *avctx;
+ AVFrame frame;
GetBitContext gb;
- DSPContext dsp;
+ int version_b; ///< Bink version 'b'
int first;
int channels;
int frame_len; ///< transform size (samples)
int num_bands;
unsigned int *bands;
float root;
- DECLARE_ALIGNED_16(FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
- DECLARE_ALIGNED_16(short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
- float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
+ DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
+ float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
+ uint8_t *packet_buffer;
union {
RDFTContext rdft;
DCTContext dct;
int i;
int frame_len_bits;
- s->avctx = avctx;
- dsputil_init(&s->dsp, avctx);
-
/* determine frame length */
if (avctx->sample_rate < 22050) {
frame_len_bits = 9;
} else {
frame_len_bits = 11;
}
- s->frame_len = 1 << frame_len_bits;
- if (s->channels > MAX_CHANNELS) {
- av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
+ if (avctx->channels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels);
return -1;
}
- if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
+ s->version_b = avctx->extradata && avctx->extradata[3] == 'b';
+
+ if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
// audio is already interleaved for the RDFT format variant
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
sample_rate *= avctx->channels;
- s->frame_len *= avctx->channels;
s->channels = 1;
- if (avctx->channels == 2)
- frame_len_bits++;
+ if (!s->version_b)
+ frame_len_bits += av_log2(avctx->channels);
} else {
s->channels = avctx->channels;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
}
+ s->frame_len = 1 << frame_len_bits;
s->overlap_len = s->frame_len / 16;
s->block_size = (s->frame_len - s->overlap_len) * s->channels;
sample_rate_half = (sample_rate + 1) / 2;
- s->root = 2.0 / sqrt(s->frame_len);
+ if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
+ s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
+ else
+ s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
+ for (i = 0; i < 96; i++) {
+ /* constant is result of 0.066399999/log10(M_E) */
+ quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
+ }
/* calculate number of bands */
for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
return AVERROR(ENOMEM);
/* populate bands data */
- s->bands[0] = 1;
+ s->bands[0] = 2;
for (i = 1; i < s->num_bands; i++)
- s->bands[i] = ff_wma_critical_freqs[i - 1] * (s->frame_len / 2) / sample_rate_half;
- s->bands[s->num_bands] = s->frame_len / 2;
+ s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
+ s->bands[s->num_bands] = s->frame_len;
s->first = 1;
- avctx->sample_fmt = SAMPLE_FMT_S16;
- for (i = 0; i < s->channels; i++)
- s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
-
- if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
- ff_rdft_init(&s->trans.rdft, frame_len_bits, IRIDFT);
+ if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
+ ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
else if (CONFIG_BINKAUDIO_DCT_DECODER)
- ff_dct_init(&s->trans.dct, frame_len_bits, 1);
+ ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
else
return -1;
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
}
/**
* Decode Bink Audio block
* @param[out] out Output buffer (must contain s->block_size elements)
+ * @return 0 on success, negative error code on failure
*/
-static void decode_block(BinkAudioContext *s, short *out, int use_dct)
+static int decode_block(BinkAudioContext *s, float **out, int use_dct)
{
int ch, i, j, k;
float q, quant[25];
skip_bits(gb, 2);
for (ch = 0; ch < s->channels; ch++) {
- FFTSample *coeffs = s->coeffs_ptr[ch];
- q = 0.0f;
- coeffs[0] = get_float(gb) * s->root;
- coeffs[1] = get_float(gb) * s->root;
+ FFTSample *coeffs = out[ch];
+
+ if (s->version_b) {
+ if (get_bits_left(gb) < 64)
+ return AVERROR_INVALIDDATA;
+ coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
+ coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
+ } else {
+ if (get_bits_left(gb) < 58)
+ return AVERROR_INVALIDDATA;
+ coeffs[0] = get_float(gb) * s->root;
+ coeffs[1] = get_float(gb) * s->root;
+ }
+ if (get_bits_left(gb) < s->num_bands * 8)
+ return AVERROR_INVALIDDATA;
for (i = 0; i < s->num_bands; i++) {
- /* constant is result of 0.066399999/log10(M_E) */
int value = get_bits(gb, 8);
- quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
+ quant[i] = quant_table[FFMIN(value, 95)];
}
- // find band (k)
- for (k = 0; s->bands[k] < 1; k++) {
- q = quant[k];
- }
+ k = 0;
+ q = quant[0];
// parse coefficients
i = 2;
while (i < s->frame_len) {
- if (get_bits1(gb)) {
- j = i + rle_length_tab[get_bits(gb, 4)] * 8;
+ if (s->version_b) {
+ j = i + 16;
} else {
- j = i + 8;
+ int v = get_bits1(gb);
+ if (v) {
+ v = get_bits(gb, 4);
+ j = i + rle_length_tab[v] * 8;
+ } else {
+ j = i + 8;
+ }
}
j = FFMIN(j, s->frame_len);
if (width == 0) {
memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
i = j;
- while (s->bands[k] * 2 < i)
+ while (s->bands[k] < i)
q = quant[k++];
} else {
while (i < j) {
- if (s->bands[k] * 2 == i)
+ if (s->bands[k] == i)
q = quant[k++];
coeff = get_bits(gb, width);
if (coeff) {
- if (get_bits1(gb))
+ int v;
+ v = get_bits1(gb);
+ if (v)
coeffs[i] = -q * coeff;
else
coeffs[i] = q * coeff;
if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
coeffs[0] /= 0.5;
- ff_dct_calc (&s->trans.dct, coeffs);
- s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
+ s->trans.dct.dct_calc(&s->trans.dct, coeffs);
}
else if (CONFIG_BINKAUDIO_RDFT_DECODER)
- ff_rdft_calc(&s->trans.rdft, coeffs);
+ s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
}
- s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels);
-
- if (!s->first) {
+ for (ch = 0; ch < s->channels; ch++) {
+ int j;
int count = s->overlap_len * s->channels;
- int shift = av_log2(count);
- for (i = 0; i < count; i++) {
- out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
+ if (!s->first) {
+ j = ch;
+ for (i = 0; i < s->overlap_len; i++, j += s->channels)
+ out[ch][i] = (s->previous[ch][i] * (count - j) +
+ out[ch][i] * j) / count;
}
+ memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
+ s->overlap_len * sizeof(*s->previous[ch]));
}
- memcpy(s->previous, out + s->block_size,
- s->overlap_len * s->channels * sizeof(*out));
-
s->first = 0;
+
+ return 0;
}
static av_cold int decode_end(AVCodecContext *avctx)
{
BinkAudioContext * s = avctx->priv_data;
av_freep(&s->bands);
- if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
+ av_freep(&s->packet_buffer);
+ if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_end(&s->trans.rdft);
else if (CONFIG_BINKAUDIO_DCT_DECODER)
ff_dct_end(&s->trans.dct);
+
return 0;
}
if (n) skip_bits(s, n);
}
-static int decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
BinkAudioContext *s = avctx->priv_data;
- const uint8_t *buf = avpkt->data;
- int buf_size = avpkt->size;
- short *samples = data;
- short *samples_end = (short*)((uint8_t*)data + *data_size);
- int reported_size;
GetBitContext *gb = &s->gb;
+ int ret, consumed = 0;
+
+ if (!get_bits_left(gb)) {
+ uint8_t *buf;
+ /* handle end-of-stream */
+ if (!avpkt->size) {
+ *got_frame_ptr = 0;
+ return 0;
+ }
+ if (avpkt->size < 4) {
+ av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
+ return AVERROR_INVALIDDATA;
+ }
+ buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!buf)
+ return AVERROR(ENOMEM);
+ s->packet_buffer = buf;
+ memcpy(s->packet_buffer, avpkt->data, avpkt->size);
+ init_get_bits(gb, s->packet_buffer, avpkt->size * 8);
+ consumed = avpkt->size;
+
+ /* skip reported size */
+ skip_bits_long(gb, 32);
+ }
- init_get_bits(gb, buf, buf_size * 8);
+ /* get output buffer */
+ s->frame.nb_samples = s->frame_len;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
- reported_size = get_bits_long(gb, 32);
- while (get_bits_count(gb) / 8 < buf_size &&
- samples + s->block_size <= samples_end) {
- decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
- samples += s->block_size;
- get_bits_align32(gb);
+ if (decode_block(s, (float **)s->frame.extended_data,
+ avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
+ av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
+ return AVERROR_INVALIDDATA;
}
+ get_bits_align32(gb);
+
+ s->frame.nb_samples = s->block_size / avctx->channels;
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
- *data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
- return buf_size;
+ return consumed;
}
-AVCodec binkaudio_rdft_decoder = {
- "binkaudio_rdft",
- CODEC_TYPE_AUDIO,
- CODEC_ID_BINKAUDIO_RDFT,
- sizeof(BinkAudioContext),
- decode_init,
- NULL,
- decode_end,
- decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
+AVCodec ff_binkaudio_rdft_decoder = {
+ .name = "binkaudio_rdft",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_BINKAUDIO_RDFT,
+ .priv_data_size = sizeof(BinkAudioContext),
+ .init = decode_init,
+ .close = decode_end,
+ .decode = decode_frame,
+ .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
};
-AVCodec binkaudio_dct_decoder = {
- "binkaudio_dct",
- CODEC_TYPE_AUDIO,
- CODEC_ID_BINKAUDIO_DCT,
- sizeof(BinkAudioContext),
- decode_init,
- NULL,
- decode_end,
- decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
+AVCodec ff_binkaudio_dct_decoder = {
+ .name = "binkaudio_dct",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_BINKAUDIO_DCT,
+ .priv_data_size = sizeof(BinkAudioContext),
+ .init = decode_init,
+ .close = decode_end,
+ .decode = decode_frame,
+ .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
};
projects / ffmpeg / blobdiff