[Ffmpeg-cvslog] r8387 - trunk/libavcodec/cook.c
banan
subversion
Tue Mar 13 20:44:02 CET 2007
Author: banan
Date: Tue Mar 13 20:44:02 2007
New Revision: 8387
Modified:
trunk/libavcodec/cook.c
Log:
Combine the window and overlap loops,
and get rid of the data swapping.
Patch by Ian Braithwaite <dk dot braithwaite at ian>
Modified: trunk/libavcodec/cook.c
==============================================================================
--- trunk/libavcodec/cook.c (original)
+++ trunk/libavcodec/cook.c Tue Mar 13 20:44:02 2007
@@ -685,33 +685,6 @@ static void mono_decode(COOKContext *q,
/**
- * The modulated lapped transform, this takes transform coefficients
- * and transforms them into timedomain samples. This is done through
- * an FFT-based algorithm with pre- and postrotation steps.
- * A window and reorder step is also included.
- *
- * @param q pointer to the COOKContext
- * @param inbuffer pointer to the mltcoefficients
- * @param outbuffer pointer to the timedomain buffer
- * @param mlt_tmp pointer to temporary storage space
- */
-
-static void cook_imlt(COOKContext *q, float* inbuffer, float* outbuffer)
-{
- int i;
-
- q->mdct_ctx.fft.imdct_calc(&q->mdct_ctx, outbuffer, inbuffer, q->mdct_tmp);
-
- for(i = 0; i < q->samples_per_channel; i++){
- float tmp = outbuffer[i];
-
- outbuffer[i] = q->mlt_window[i] * outbuffer[q->samples_per_channel + i];
- outbuffer[q->samples_per_channel + i] = q->mlt_window[q->samples_per_channel - 1 - i] * -tmp;
- }
-}
-
-
-/**
* the actual requantization of the timedomain samples
*
* @param q pointer to the COOKContext
@@ -743,36 +716,50 @@ static void interpolate(COOKContext *q,
/**
- * mlt overlapping and buffer management
+ * The modulated lapped transform, this takes transform coefficients
+ * and transforms them into timedomain samples.
+ * Apply transform window, overlap buffers, apply gain profile
+ * and buffer management.
*
* @param q pointer to the COOKContext
+ * @param inbuffer pointer to the mltcoefficients
* @param gains_ptr current and previous gains
* @param previous_buffer pointer to the previous buffer to be used for overlapping
*/
-static void gain_compensate(COOKContext *q, cook_gains *gains_ptr,
- float* previous_buffer)
+static void imlt_gain(COOKContext *q, float *inbuffer,
+ cook_gains *gains_ptr, float* previous_buffer)
{
const float fc = q->pow2tab[gains_ptr->previous[0] + 63];
- float *buffer = q->mono_mdct_output;
+ float *buffer0 = q->mono_mdct_output;
+ float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
int i;
- /* Overlap with the previous block. */
- for(i=0 ; i<q->samples_per_channel ; i++) {
- buffer[i] *= fc;
- buffer[i] += previous_buffer[i];
+ /* Inverse modified discrete cosine transform */
+ q->mdct_ctx.fft.imdct_calc(&q->mdct_ctx, q->mono_mdct_output,
+ inbuffer, q->mdct_tmp);
+
+ /* The weird thing here, is that the two halves of the time domain
+ * buffer are swapped. Also, the newest data, that we save away for
+ * next frame, has the wrong sign. Hence the subtraction below.
+ * Almost sounds like a complex conjugate/reverse data/FFT effect.
+ */
+
+ /* Apply window and overlap */
+ for(i = 0; i < q->samples_per_channel; i++){
+ buffer1[i] = buffer1[i] * fc * q->mlt_window[i] -
+ previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
}
/* Apply gain profile */
for (i = 0; i < 8; i++) {
if (gains_ptr->now[i] || gains_ptr->now[i + 1])
- interpolate(q, &buffer[q->gain_size_factor * i],
+ interpolate(q, &buffer1[q->gain_size_factor * i],
gains_ptr->now[i], gains_ptr->now[i + 1]);
}
/* Save away the current to be previous block. */
- memcpy(previous_buffer, buffer+q->samples_per_channel,
- sizeof(float)*q->samples_per_channel);
+ memcpy(previous_buffer, buffer0, sizeof(float)*q->samples_per_channel);
}
@@ -902,16 +889,16 @@ mlt_compensate_output(COOKContext *q, fl
cook_gains *gains, float *previous_buffer,
int16_t *out, int chan)
{
+ float *output = q->mono_mdct_output + q->samples_per_channel;
int j;
- cook_imlt(q, decode_buffer, q->mono_mdct_output);
- gain_compensate(q, gains, previous_buffer);
+ imlt_gain(q, decode_buffer, gains, previous_buffer);
/* Clip and convert floats to 16 bits.
*/
for (j = 0; j < q->samples_per_channel; j++) {
out[chan + q->nb_channels * j] =
- av_clip(lrintf(q->mono_mdct_output[j]), -32768, 32767);
+ av_clip(lrintf(output[j]), -32768, 32767);
}
}
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