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binkaudio: store interleaved overlap samples in BinkAudioContext.

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This fixes the requirement for the buffer size to be larger than the number of
samples actually decoded.
This commit is contained in:
Justin Ruggles 2011-10-26 10:18:39 -04:00
parent 9f48039a37
commit eaddd29e00
1 changed files with 12 additions and 4 deletions
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16
libavcodec/binkaudio.c
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@ -59,7 +59,9 @@ typedef struct {
float root; float root;
DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE]; 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, short, 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 *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
union { union {
RDFTContext rdft; RDFTContext rdft;
DCTContext dct; DCTContext dct;
@ -132,8 +134,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
s->first = 1; s->first = 1;
avctx->sample_fmt = AV_SAMPLE_FMT_S16; 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->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) if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R); ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
@ -256,8 +260,12 @@ static int decode_block(BinkAudioContext *s, short *out, int use_dct)
s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs); 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->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) { if (!s->first) {
int count = s->overlap_len * s->channels; int count = s->overlap_len * s->channels;
@ -267,8 +275,8 @@ static int decode_block(BinkAudioContext *s, short *out, int use_dct)
} }
} }
memcpy(s->previous, out + s->block_size, memcpy(s->previous, s->current,
s->overlap_len * s->channels * sizeof(*out)); s->overlap_len * s->channels * sizeof(*s->previous));
s->first = 0; s->first = 0;