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binkaudio: use float sample format

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Use planar for DCT codec, interleaved for RDFT codec.
This commit is contained in:
Justin Ruggles 2012-08-26 20:41:45 -04:00
parent 0801b5979b
commit 7bfd1766d1
1 changed files with 21 additions and 35 deletions
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56
libavcodec/binkaudio.c
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@ -47,7 +47,6 @@ static float quant_table[96];
typedef struct {
AVFrame frame;
GetBitContext gb;
FmtConvertContext fmt_conv;
int version_b; ///< Bink version 'b'
int first;
int channels;
@ -58,10 +57,7 @@ typedef struct {
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;
@ -78,8 +74,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
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;
@ -98,12 +92,14 @@ static av_cold int decode_init(AVCodecContext *avctx)
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;
@ -111,9 +107,9 @@ static av_cold int decode_init(AVCodecContext *avctx)
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;
@ -135,12 +131,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
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);
@ -179,7 +169,7 @@ static const uint8_t rle_length_tab[16] = {
* @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];
@ -190,7 +180,8 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
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;
@ -265,24 +256,19 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
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;
@ -311,7 +297,6 @@ static int decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
BinkAudioContext *s = avctx->priv_data;
int16_t *samples;
GetBitContext *gb = &s->gb;
int ret, consumed = 0;
@ -339,19 +324,20 @@ static int decode_frame(AVCodecContext *avctx, void *data,
}
/* 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;