wmavoice: convert RDFT to lavu/tx
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@ -31,6 +31,7 @@
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#include "libavutil/float_dsp.h"
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#include "libavutil/mem_internal.h"
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#include "libavutil/thread.h"
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#include "libavutil/tx.h"
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#include "avcodec.h"
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#include "codec_internal.h"
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#include "decode.h"
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@ -263,8 +264,8 @@ typedef struct WMAVoiceContext {
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* smoothing and so on, and context variables for FFT/iFFT.
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* @{
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*/
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RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the
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///< postfilter (for denoise filter)
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AVTXContext *rdft, *irdft; ///< contexts for FFT-calculation in the
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av_tx_fn rdft_fn, irdft_fn; ///< postfilter (for denoise filter)
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DCTContext dct, dst; ///< contexts for phase shift (in Hilbert
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///< transform, part of postfilter)
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float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi]
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@ -277,9 +278,9 @@ typedef struct WMAVoiceContext {
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///< by postfilter
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float denoise_filter_cache[MAX_FRAMESIZE];
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int denoise_filter_cache_size; ///< samples in #denoise_filter_cache
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DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
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DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x82];
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///< aligned buffer for LPC tilting
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DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
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DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x82];
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///< aligned buffer for denoise coefficients
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DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
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///< aligned buffer for postfilter speech
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@ -388,12 +389,20 @@ static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
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s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
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s->do_apf = flags & 0x1;
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if (s->do_apf) {
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if ((ret = ff_rdft_init(&s->rdft, 7, DFT_R2C)) < 0 ||
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(ret = ff_rdft_init(&s->irdft, 7, IDFT_C2R)) < 0 ||
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(ret = ff_dct_init (&s->dct, 6, DCT_I)) < 0 ||
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float scale = 1.0f;
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if ((ret = ff_dct_init (&s->dct, 6, DCT_I)) < 0 ||
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(ret = ff_dct_init (&s->dst, 6, DST_I)) < 0)
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return ret;
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ret = av_tx_init(&s->rdft, &s->rdft_fn, AV_TX_FLOAT_RDFT, 0, 1 << 7, &scale, 0);
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if (ret < 0)
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return ret;
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ret = av_tx_init(&s->irdft, &s->irdft_fn, AV_TX_FLOAT_RDFT, 1, 1 << 7, &scale, 0);
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if (ret < 0)
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return ret;
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ff_sine_window_init(s->cos, 256);
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memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
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for (n = 0; n < 255; n++) {
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@ -596,20 +605,24 @@ static float tilt_factor(const float *lpcs, int n_lpcs)
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/**
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* Derive denoise filter coefficients (in real domain) from the LPCs.
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*/
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static void calc_input_response(WMAVoiceContext *s, float *lpcs,
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int fcb_type, float *coeffs, int remainder)
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static void calc_input_response(WMAVoiceContext *s, float *lpcs_src,
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int fcb_type, float *coeffs_dst, int remainder)
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{
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float last_coeff, min = 15.0, max = -15.0;
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float irange, angle_mul, gain_mul, range, sq;
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LOCAL_ALIGNED_32(float, coeffs, [0x82]);
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LOCAL_ALIGNED_32(float, lpcs, [0x82]);
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int n, idx;
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memcpy(coeffs, coeffs_dst, 0x82*sizeof(float));
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/* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
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s->rdft.rdft_calc(&s->rdft, lpcs);
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s->rdft_fn(s->rdft, lpcs, lpcs_src, sizeof(float));
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#define log_range(var, assign) do { \
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float tmp = log10f(assign); var = tmp; \
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max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
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} while (0)
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log_range(last_coeff, lpcs[1] * lpcs[1]);
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log_range(last_coeff, lpcs[64] * lpcs[64]);
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for (n = 1; n < 64; n++)
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log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] +
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lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
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@ -668,25 +681,25 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
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coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
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coeffs[n * 2] = coeffs[n] * s->cos[idx];
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}
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coeffs[1] = last_coeff;
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coeffs[64] = last_coeff;
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/* move into real domain */
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s->irdft.rdft_calc(&s->irdft, coeffs);
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s->irdft_fn(s->irdft, coeffs_dst, coeffs, sizeof(AVComplexFloat));
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/* tilt correction and normalize scale */
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memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
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memset(&coeffs_dst[remainder], 0, sizeof(coeffs_dst[0]) * (128 - remainder));
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if (s->denoise_tilt_corr) {
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float tilt_mem = 0;
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coeffs[remainder - 1] = 0;
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coeffs_dst[remainder - 1] = 0;
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ff_tilt_compensation(&tilt_mem,
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-1.8 * tilt_factor(coeffs, remainder - 1),
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coeffs, remainder);
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-1.8 * tilt_factor(coeffs_dst, remainder - 1),
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coeffs_dst, remainder);
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}
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sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
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sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs_dst, coeffs_dst,
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remainder));
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for (n = 0; n < remainder; n++)
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coeffs[n] *= sq;
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coeffs_dst[n] *= sq;
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}
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/**
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@ -722,6 +735,8 @@ static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
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int remainder, lim, n;
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if (fcb_type != FCB_TYPE_SILENCE) {
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LOCAL_ALIGNED_32(float, coeffs_f, [0x82]);
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LOCAL_ALIGNED_32(float, synth_f, [0x82]);
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float *tilted_lpcs = s->tilted_lpcs_pf,
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*coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
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@ -742,16 +757,16 @@ static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
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/* apply coefficients (in frequency spectrum domain), i.e. complex
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* number multiplication */
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memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
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s->rdft.rdft_calc(&s->rdft, synth_pf);
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s->rdft.rdft_calc(&s->rdft, coeffs);
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synth_pf[0] *= coeffs[0];
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synth_pf[1] *= coeffs[1];
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for (n = 1; n < 64; n++) {
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float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
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synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
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synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
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s->rdft_fn(s->rdft, synth_f, synth_pf, sizeof(float));
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s->rdft_fn(s->rdft, coeffs_f, coeffs, sizeof(float));
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synth_f[0] *= coeffs_f[0];
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synth_f[1] *= coeffs_f[1];
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for (n = 1; n <= 64; n++) {
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float v1 = synth_f[n * 2], v2 = synth_f[n * 2 + 1];
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synth_f[n * 2] = v1 * coeffs_f[n * 2] - v2 * coeffs_f[n * 2 + 1];
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synth_f[n * 2 + 1] = v2 * coeffs_f[n * 2] + v1 * coeffs_f[n * 2 + 1];
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}
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s->irdft.rdft_calc(&s->irdft, synth_pf);
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s->irdft_fn(s->irdft, synth_pf, synth_f, sizeof(AVComplexFloat));
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}
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/* merge filter output with the history of previous runs */
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@ -1986,8 +2001,8 @@ static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
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WMAVoiceContext *s = ctx->priv_data;
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if (s->do_apf) {
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ff_rdft_end(&s->rdft);
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ff_rdft_end(&s->irdft);
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av_tx_uninit(&s->rdft);
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av_tx_uninit(&s->irdft);
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ff_dct_end(&s->dct);
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ff_dct_end(&s->dst);
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}
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