* Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
* Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
*
- * 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
*/
typedef struct {
AVFrame frame;
GetBitContext gb;
- FmtConvertContext fmt_conv;
int version_b; ///< Bink version 'b'
int first;
int channels;
unsigned int *bands;
float root;
DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
- DECLARE_ALIGNED(16, int16_t, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
- DECLARE_ALIGNED(16, int16_t, current)[BINK_BLOCK_MAX_SIZE / 16];
- float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
- float *prev_ptr[MAX_CHANNELS]; ///< pointers to the overlap points in the coeffs array
+ float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
uint8_t *packet_buffer;
union {
RDFTContext rdft;
int i;
int frame_len_bits;
- ff_fmt_convert_init(&s->fmt_conv, avctx);
-
/* determine frame length */
if (avctx->sample_rate < 22050) {
frame_len_bits = 9;
frame_len_bits = 11;
}
- if (avctx->channels > MAX_CHANNELS) {
- av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels);
- return -1;
+ if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);
+ return AVERROR_INVALIDDATA;
}
- s->version_b = avctx->extradata && avctx->extradata[3] == 'b';
+ s->version_b = avctx->extradata_size >= 4 && 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->channels = 1;
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->block_size = (s->frame_len - s->overlap_len) * s->channels;
sample_rate_half = (sample_rate + 1) / 2;
if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
- s->root = 2.0 / sqrt(s->frame_len);
+ s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
else
- s->root = s->frame_len / sqrt(s->frame_len);
+ 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;
s->bands[s->num_bands] = s->frame_len;
s->first = 1;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
-
- for (i = 0; i < s->channels; i++) {
- s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
- s->prev_ptr[i] = s->coeffs_ptr[i] + s->frame_len - s->overlap_len;
- }
if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
};
-#define GET_BITS_SAFE(out, nbits) do { \
- if (get_bits_left(gb) < nbits) \
- return AVERROR_INVALIDDATA; \
- out = get_bits(gb, nbits); \
-} while (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 int decode_block(BinkAudioContext *s, int16_t *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];
+ FFTSample *coeffs = out[ch];
+
if (s->version_b) {
if (get_bits_left(gb) < 64)
return AVERROR_INVALIDDATA;
if (s->version_b) {
j = i + 16;
} else {
- int v;
- GET_BITS_SAFE(v, 1);
+ int v = get_bits1(gb);
if (v) {
- GET_BITS_SAFE(v, 4);
+ v = get_bits(gb, 4);
j = i + rle_length_tab[v] * 8;
} else {
j = i + 8;
j = FFMIN(j, s->frame_len);
- GET_BITS_SAFE(width, 4);
+ width = get_bits(gb, 4);
if (width == 0) {
memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
i = j;
while (i < j) {
if (s->bands[k] == i)
q = quant[k++];
- GET_BITS_SAFE(coeff, width);
+ coeff = get_bits(gb, width);
if (coeff) {
int v;
- GET_BITS_SAFE(v, 1);
+ v = get_bits1(gb);
if (v)
coeffs[i] = -q * coeff;
else
s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
}
- s->fmt_conv.float_to_int16_interleave(s->current,
- (const float **)s->prev_ptr,
- s->overlap_len, s->channels);
- s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
- s->frame_len - s->overlap_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, s->current,
- s->overlap_len * s->channels * sizeof(*s->previous));
-
s->first = 0;
return 0;
int *got_frame_ptr, AVPacket *avpkt)
{
BinkAudioContext *s = avctx->priv_data;
- int16_t *samples;
GetBitContext *gb = &s->gb;
int ret, consumed = 0;
}
/* get output buffer */
- s->frame.nb_samples = s->block_size / avctx->channels;
+ 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;
}
- samples = (int16_t *)s->frame.data[0];
- if (decode_block(s, samples, avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
+ 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;
projects / ffmpeg / blobdiff