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aacenc: remove the data arrays
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 2790d7a9ff
)
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c0fd48a702
commit
5e06b02005
@ -225,40 +225,41 @@ static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
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const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
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const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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float *output = sce->ret;
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if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
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memcpy(s->output, sce->saved, sizeof(float)*1024);
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memcpy(output, sce->saved, sizeof(float)*1024);
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if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
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memset(s->output, 0, sizeof(s->output[0]) * 448);
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memset(output, 0, sizeof(output[0]) * 448);
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for (i = 448; i < 576; i++)
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s->output[i] = sce->saved[i] * pwindow[i - 448];
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output[i] = sce->saved[i] * pwindow[i - 448];
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for (i = 576; i < 704; i++)
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s->output[i] = sce->saved[i];
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output[i] = sce->saved[i];
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}
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if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
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for (i = 0; i < 1024; i++) {
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s->output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
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output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
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sce->saved[i] = audio[i * chans] * lwindow[i];
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}
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} else {
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for (i = 0; i < 448; i++)
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s->output[i+1024] = audio[i * chans];
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output[i+1024] = audio[i * chans];
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for (; i < 576; i++)
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s->output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
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memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
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output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
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memset(output+1024+576, 0, sizeof(output[0]) * 448);
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for (i = 0; i < 1024; i++)
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sce->saved[i] = audio[i * chans];
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}
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ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
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ff_mdct_calc(&s->mdct1024, sce->coeffs, output);
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} else {
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for (k = 0; k < 1024; k += 128) {
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for (i = 448 + k; i < 448 + k + 256; i++)
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s->output[i - 448 - k] = (i < 1024)
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output[i - 448 - k] = (i < 1024)
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? sce->saved[i]
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: audio[(i-1024)*chans];
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s->dsp.vector_fmul (s->output, s->output, k ? swindow : pwindow, 128);
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s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
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ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
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s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128);
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s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
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ff_mdct_calc(&s->mdct128, sce->coeffs + k, output);
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}
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for (i = 0; i < 1024; i++)
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sce->saved[i] = audio[i * chans];
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@ -597,6 +598,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
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}
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for (j = 0; j < chans; j++) {
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s->cur_channel = start_ch + j;
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s->scoefs = cpe->ch[j].ret;
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encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
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}
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start_ch += chans;
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@ -52,8 +52,7 @@ typedef struct AACEncContext {
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FFTContext mdct1024; ///< long (1024 samples) frame transform context
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FFTContext mdct128; ///< short (128 samples) frame transform context
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DSPContext dsp;
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DECLARE_ALIGNED(16, FFTSample, output)[2048]; ///< temporary buffer for MDCT input coefficients
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int16_t* samples; ///< saved preprocessed input
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int16_t *samples; ///< saved preprocessed input
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int samplerate_index; ///< MPEG-4 samplerate index
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@ -64,8 +63,8 @@ typedef struct AACEncContext {
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int cur_channel;
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int last_frame;
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float lambda;
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float *scoefs; ///< scaled coefficients
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DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
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DECLARE_ALIGNED(16, float, scoefs)[1024]; ///< scaled coefficients
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} AACEncContext;
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#endif /* AVCODEC_AACENC_H */
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