提交 2790d7a9 编写于 作者: Y Young Han Lee 提交者: Ronald S. Bultje

aacenc: remove the data arrays

Signed-off-by: NRonald S. Bultje <rsbultje@gmail.com>
上级 c9256246
......@@ -225,40 +225,41 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
float *output = sce->ret;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
memcpy(s->output, sce->saved, sizeof(float)*1024);
memcpy(output, sce->saved, sizeof(float)*1024);
if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
memset(s->output, 0, sizeof(s->output[0]) * 448);
memset(output, 0, sizeof(output[0]) * 448);
for (i = 448; i < 576; i++)
s->output[i] = sce->saved[i] * pwindow[i - 448];
output[i] = sce->saved[i] * pwindow[i - 448];
for (i = 576; i < 704; i++)
s->output[i] = sce->saved[i];
output[i] = sce->saved[i];
}
if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
for (i = 0; i < 1024; i++) {
s->output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
sce->saved[i] = audio[i * chans] * lwindow[i];
}
} else {
for (i = 0; i < 448; i++)
s->output[i+1024] = audio[i * chans];
output[i+1024] = audio[i * chans];
for (; i < 576; i++)
s->output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
memset(output+1024+576, 0, sizeof(output[0]) * 448);
for (i = 0; i < 1024; i++)
sce->saved[i] = audio[i * chans];
}
ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
ff_mdct_calc(&s->mdct1024, sce->coeffs, output);
} else {
for (k = 0; k < 1024; k += 128) {
for (i = 448 + k; i < 448 + k + 256; i++)
s->output[i - 448 - k] = (i < 1024)
output[i - 448 - k] = (i < 1024)
? sce->saved[i]
: audio[(i-1024)*chans];
s->dsp.vector_fmul (s->output, s->output, k ? swindow : pwindow, 128);
s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128);
s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
ff_mdct_calc(&s->mdct128, sce->coeffs + k, output);
}
for (i = 0; i < 1024; i++)
sce->saved[i] = audio[i * chans];
......@@ -597,6 +598,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
}
for (j = 0; j < chans; j++) {
s->cur_channel = start_ch + j;
s->scoefs = cpe->ch[j].ret;
encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
}
start_ch += chans;
......
......@@ -52,8 +52,7 @@ typedef struct AACEncContext {
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
DSPContext dsp;
DECLARE_ALIGNED(16, FFTSample, output)[2048]; ///< temporary buffer for MDCT input coefficients
int16_t* samples; ///< saved preprocessed input
int16_t *samples; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
......@@ -64,8 +63,8 @@ typedef struct AACEncContext {
int cur_channel;
int last_frame;
float lambda;
float *scoefs; ///< scaled coefficients
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(16, float, scoefs)[1024]; ///< scaled coefficients
} AACEncContext;
#endif /* AVCODEC_AACENC_H */
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