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 {
+ AVFrame frame;
GetBitContext gb;
DSPContext dsp;
FmtConvertContext fmt_conv;
unsigned int *bands;
float root;
DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
- DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
+ 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
+ uint8_t *packet_buffer;
union {
RDFTContext rdft;
DCTContext dct;
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);
+ 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++)
s->first = 1;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- for (i = 0; i < s->channels; i++)
+ 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 == CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
else
return -1;
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
}
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 void decode_block(BinkAudioContext *s, short *out, int use_dct)
+static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
{
int ch, i, j, k;
float q, quant[25];
for (ch = 0; ch < s->channels; ch++) {
FFTSample *coeffs = s->coeffs_ptr[ch];
if (s->version_b) {
+ if (get_bits_left(gb) < 64)
+ return AVERROR_INVALIDDATA;
coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
coeffs[1] = av_int2flt(get_bits(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)];
}
k = 0;
while (i < s->frame_len) {
if (s->version_b) {
j = i + 16;
- } else if (get_bits1(gb)) {
- j = i + rle_length_tab[get_bits(gb, 4)] * 8;
} else {
- j = i + 8;
+ int v;
+ GET_BITS_SAFE(v, 1);
+ if (v) {
+ GET_BITS_SAFE(v, 4);
+ j = i + rle_length_tab[v] * 8;
+ } else {
+ j = i + 8;
+ }
}
j = FFMIN(j, s->frame_len);
- width = get_bits(gb, 4);
+ GET_BITS_SAFE(width, 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++];
- coeff = get_bits(gb, width);
+ GET_BITS_SAFE(coeff, width);
if (coeff) {
- if (get_bits1(gb))
+ int v;
+ GET_BITS_SAFE(v, 1);
+ if (v)
coeffs[i] = -q * coeff;
else
coeffs[i] = q * coeff;
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->channels);
+ s->frame_len - s->overlap_len,
+ s->channels);
if (!s->first) {
int count = s->overlap_len * s->channels;
}
}
- memcpy(s->previous, out + s->block_size,
- s->overlap_len * s->channels * sizeof(*out));
+ memcpy(s->previous, s->current,
+ s->overlap_len * s->channels * sizeof(*s->previous));
s->first = 0;
+
+ return 0;
}
static av_cold int decode_end(AVCodecContext *avctx)
{
BinkAudioContext * s = avctx->priv_data;
av_freep(&s->bands);
+ av_freep(&s->packet_buffer);
if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == 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;
+ int16_t *samples;
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->block_size / avctx->channels;
+ 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];
- 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, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT)) {
+ av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
+ return AVERROR_INVALIDDATA;
}
+ get_bits_align32(gb);
+
+ *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 ff_binkaudio_rdft_decoder = {
.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)")
};
.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