*/
/**
- * @file libavcodec/binkaudio.c
+ * @file
* Bink Audio decoder
*
* Technical details here:
#define ALT_BITSTREAM_READER_LE
#include "get_bits.h"
#include "dsputil.h"
+#include "fft.h"
+#include "fmtconvert.h"
+
extern const uint16_t ff_wma_critical_freqs[25];
#define MAX_CHANNELS 2
#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
typedef struct {
- AVCodecContext *avctx;
GetBitContext gb;
DSPContext dsp;
+ FmtConvertContext fmt_conv;
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
+ 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
union {
RDFTContext rdft;
int i;
int frame_len_bits;
- s->avctx = avctx;
dsputil_init(&s->dsp, avctx);
+ ff_fmt_convert_init(&s->fmt_conv, avctx);
/* determine frame length */
if (avctx->sample_rate < 22050) {
}
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;
}
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;
+ avctx->sample_fmt = AV_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);
+ 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, 0);
+ ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
else
return -1;
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 (CONFIG_BINKAUDIO_DCT_DECODER && use_dct)
+ 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);
+ }
else if (CONFIG_BINKAUDIO_RDFT_DECODER)
ff_rdft_calc(&s->trans.rdft, coeffs);
}
- s->dsp.float_to_int16_interleave(out, (const float **)s->coeffs_ptr, s->frame_len, s->channels);
+ s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
+ s->frame_len, s->channels);
if (!s->first) {
int count = s->overlap_len * s->channels;
get_bits_align32(gb);
}
- *data_size = (uint8_t*)samples - (uint8_t*)data;
- if (reported_size != *data_size) {
- av_log(avctx, AV_LOG_WARNING, "reported data size (%d) does not match output data size (%d)\n",
- reported_size, *data_size);
- }
+ *data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
return buf_size;
}
-AVCodec binkaudio_rdft_decoder = {
+AVCodec ff_binkaudio_rdft_decoder = {
"binkaudio_rdft",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_BINKAUDIO_RDFT,
sizeof(BinkAudioContext),
decode_init,
.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
};
-AVCodec binkaudio_dct_decoder = {
+AVCodec ff_binkaudio_dct_decoder = {
"binkaudio_dct",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_BINKAUDIO_DCT,
sizeof(BinkAudioContext),
decode_init,