qdm2: fix a dubious pointer cast

This reworks a loop to get rid of an ugly pointer cast,
fixing errors seen with the PathScale ENZO compiler.

Signed-off-by: Mans Rullgard <mans@mansr.com>
This commit is contained in:
Mans Rullgard 2012-04-13 17:43:54 +01:00
parent 680097cb6d
commit f5be7958e3

View File

@ -140,7 +140,6 @@ typedef struct {
/// Parameters built from header parameters, do not change during playback
int group_order; ///< order of frame group
int fft_order; ///< order of FFT (actually fftorder+1)
int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
int frame_size; ///< size of data frame
int frequency_range;
int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
@ -1591,13 +1590,17 @@ static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
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]);
/* add samples to output buffer */
for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
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;
out += 2 * q->channels;
}
}
@ -1672,7 +1675,6 @@ static void dump_context(QDM2Context *q)
PRINT("checksum_size",q->checksum_size);
PRINT("channels",q->channels);
PRINT("nb_channels",q->nb_channels);
PRINT("fft_frame_size",q->fft_frame_size);
PRINT("fft_size",q->fft_size);
PRINT("sub_sampling",q->sub_sampling);
PRINT("fft_order",q->fft_order);
@ -1827,7 +1829,6 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
}
s->fft_order = av_log2(s->fft_size) + 1;
s->fft_frame_size = 2 * s->fft_size; // complex has two floats
// something like max decodable tones
s->group_order = av_log2(s->group_size) + 1;