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qdm2: convert to lavu/tx

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This commit is contained in:
Lynne 2022-11-05 15:39:10 +01:00
parent e6afa61be9
commit 1aeedd277a
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GPG Key ID: A2FEA5F03F034464
2 changed files with 27 additions and 21 deletions
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2
configure vendored
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@ -2920,7 +2920,7 @@ prores_encoder_select="fdctdsp"
prores_aw_encoder_select="fdctdsp"
prores_ks_encoder_select="fdctdsp"
qcelp_decoder_select="lsp"
qdm2_decoder_select="rdft mpegaudiodsp"
qdm2_decoder_select="mpegaudiodsp"
ra_144_decoder_select="audiodsp"
ra_144_encoder_select="audio_frame_queue lpc audiodsp"
ralf_decoder_select="golomb"

46
libavcodec/qdm2.c
View File

@ -37,6 +37,7 @@
#include "libavutil/channel_layout.h"
#include "libavutil/mem_internal.h"
#include "libavutil/thread.h"
#include "libavutil/tx.h"
#define BITSTREAM_READER_LE
#include "avcodec.h"
@ -46,7 +47,6 @@
#include "decode.h"
#include "mpegaudio.h"
#include "mpegaudiodsp.h"
#include "rdft.h"
#include "qdm2_tablegen.h"
@ -96,14 +96,9 @@ typedef struct QDM2SubPNode {
struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
} QDM2SubPNode;
typedef struct QDM2Complex {
float re;
float im;
} QDM2Complex;
typedef struct FFTTone {
float level;
QDM2Complex *complex;
AVComplexFloat *complex;
const float *table;
int phase;
int phase_shift;
@ -121,7 +116,8 @@ typedef struct FFTCoefficient {
} FFTCoefficient;
typedef struct QDM2FFT {
DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
DECLARE_ALIGNED(32, AVComplexFloat, complex)[MPA_MAX_CHANNELS][256 + 1];
DECLARE_ALIGNED(32, AVComplexFloat, temp)[MPA_MAX_CHANNELS][256];
} QDM2FFT;
/**
@ -161,7 +157,8 @@ typedef struct QDM2Context {
int fft_coefs_min_index[5];
int fft_coefs_max_index[5];
int fft_level_exp[6];
RDFTContext rdft_ctx;
AVTXContext *rdft_ctx;
av_tx_fn rdft_fn;
QDM2FFT fft;
/// I/O data
@ -1428,7 +1425,7 @@ static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
{
float level, f[6];
int i;
QDM2Complex c;
AVComplexFloat c;
const double iscale = 2.0 * M_PI / 512.0;
tone->phase += tone->phase_shift;
@ -1476,7 +1473,7 @@ static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
const double iscale = 0.25 * M_PI;
for (ch = 0; ch < q->channels; ch++) {
memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
memset(q->fft.complex[ch], 0, q->fft_size * sizeof(AVComplexFloat));
}
@ -1484,7 +1481,7 @@ static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
if (q->fft_coefs_min_index[4] >= 0)
for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
float level;
QDM2Complex c;
AVComplexFloat c;
if (q->fft_coefs[i].sub_packet != sub_packet)
break;
@ -1545,14 +1542,19 @@ static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
{
const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
float *out = q->output_buffer + channel;
int i;
q->fft.complex[channel][0].re *= 2.0f;
q->fft.complex[channel][0].im = 0.0f;
q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
q->fft.complex[channel][q->fft_size].re = 0.0f;
q->fft.complex[channel][q->fft_size].im = 0.0f;
q->rdft_fn(q->rdft_ctx, q->fft.temp[channel], q->fft.complex[channel],
sizeof(AVComplexFloat));
/* add samples to output buffer */
for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
out[0] += q->fft.complex[channel][i].re * gain;
out[q->channels] += q->fft.complex[channel][i].im * gain;
for (int i = 0; i < FFALIGN(q->fft_size, 8); i++) {
out[0] += q->fft.temp[channel][i].re * gain;
out[q->channels] += q->fft.temp[channel][i].im * gain;
out += 2 * q->channels;
}
}
@ -1613,7 +1615,8 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
{
static AVOnce init_static_once = AV_ONCE_INIT;
QDM2Context *s = avctx->priv_data;
int tmp_val, tmp, size;
int ret, tmp_val, tmp, size;
float scale = 1.0f / 2.0f;
GetByteContext gb;
/* extradata parsing
@ -1756,7 +1759,10 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
return AVERROR_INVALIDDATA;
}
ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
ret = av_tx_init(&s->rdft_ctx, &s->rdft_fn, AV_TX_FLOAT_RDFT, 1, 2*s->fft_size, &scale, 0);
if (ret < 0)
return ret;
ff_mpadsp_init(&s->mpadsp);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
@ -1770,7 +1776,7 @@ static av_cold int qdm2_decode_close(AVCodecContext *avctx)
{
QDM2Context *s = avctx->priv_data;
ff_rdft_end(&s->rdft_ctx);
av_tx_uninit(&s->rdft_ctx);
return 0;
}