Samples should always be 32 bit on all platforms, so change most
occurences of long to int32_t to enable working sounds also on 64 bit sims. Note that some codecs (MP3 and Wavpack) still have other 64 bit related problems. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9120 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Thom Johansen
parent
f383cc14bd
commit
ea4ccb5aba
3 changed files with 56 additions and 53 deletions
98
apps/dsp.c
98
apps/dsp.c
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@ -192,7 +192,7 @@ struct dsp_config
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struct resample_data
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{
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long phase, delta;
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long last_sample[2];
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int32_t last_sample[2];
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};
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struct dither_data
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@ -203,9 +203,9 @@ struct dither_data
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struct crossfeed_data
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{
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long lowpass[2];
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long highpass[2];
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long delay[2][13];
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int32_t lowpass[2];
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int32_t highpass[2];
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int32_t delay[2][13];
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int index;
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};
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@ -233,8 +233,8 @@ struct dsp_config *dsp;
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* of copying needed is minimized for that case.
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*/
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static long sample_buf[SAMPLE_BUF_SIZE] IBSS_ATTR;
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static long resample_buf[RESAMPLE_BUF_SIZE] IBSS_ATTR;
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static int32_t sample_buf[SAMPLE_BUF_SIZE] IBSS_ATTR;
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static int32_t resample_buf[RESAMPLE_BUF_SIZE] IBSS_ATTR;
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int sound_get_pitch(void)
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{
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@ -254,7 +254,7 @@ void sound_set_pitch(int permille)
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* consume. Note that for mono, dst[0] equals dst[1], as there is no point
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* in processing the same data twice.
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*/
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static int convert_to_internal(const char* src[], int count, long* dst[])
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static int convert_to_internal(const char* src[], int count, int32_t* dst[])
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{
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count = MIN(SAMPLE_BUF_SIZE / 2, count);
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@ -267,15 +267,15 @@ static int convert_to_internal(const char* src[], int count, long* dst[])
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}
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else
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{
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dst[0] = (long*) src[0];
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dst[1] = (long*) ((dsp->stereo_mode == STEREO_MONO) ? src[0] : src[1]);
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dst[0] = (int32_t*) src[0];
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dst[1] = (int32_t*) ((dsp->stereo_mode == STEREO_MONO) ? src[0] : src[1]);
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}
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if (dsp->sample_depth <= NATIVE_DEPTH)
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{
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short* s0 = (short*) src[0];
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long* d0 = dst[0];
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long* d1 = dst[1];
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int32_t* d0 = dst[0];
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int32_t* d1 = dst[1];
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int scale = WORD_SHIFT;
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int i;
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@ -307,9 +307,9 @@ static int convert_to_internal(const char* src[], int count, long* dst[])
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}
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else if (dsp->stereo_mode == STEREO_INTERLEAVED)
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{
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long* s0 = (long*) src[0];
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long* d0 = dst[0];
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long* d1 = dst[1];
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int32_t* s0 = (int32_t*) src[0];
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int32_t* d0 = dst[0];
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int32_t* d1 = dst[1];
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int i;
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for (i = 0; i < count; i++)
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@ -348,13 +348,13 @@ static void resampler_set_delta(int frequency)
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/* TODO: we really should have a separate set of resample functions for both
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mono and stereo to avoid all this internal branching and looping. */
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static long downsample(long **dst, long **src, int count,
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static long downsample(int32_t **dst, int32_t **src, int count,
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struct resample_data *r)
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{
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long phase = r->phase;
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long delta = r->delta;
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long last_sample;
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long *d[2] = { dst[0], dst[1] };
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int32_t last_sample;
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int32_t *d[2] = { dst[0], dst[1] };
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int pos = phase >> 16;
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int i = 1, j;
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int num_channels = dsp->stereo_mode == STEREO_MONO ? 1 : 2;
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@ -388,11 +388,11 @@ static long downsample(long **dst, long **src, int count,
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return i;
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}
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static long upsample(long **dst, long **src, int count, struct resample_data *r)
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static long upsample(int32_t **dst, int32_t **src, int count, struct resample_data *r)
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{
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long phase = r->phase;
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long delta = r->delta;
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long *d[2] = { dst[0], dst[1] };
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int32_t *d[2] = { dst[0], dst[1] };
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int i = 0, j;
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int pos;
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int num_channels = dsp->stereo_mode == STEREO_MONO ? 1 : 2;
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@ -427,13 +427,13 @@ static long upsample(long **dst, long **src, int count, struct resample_data *r)
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* done, to refer to the resampled data. Returns number of stereo samples
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* for further processing.
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*/
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static inline int resample(long* src[], int count)
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static inline int resample(int32_t* src[], int count)
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{
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long new_count;
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if (dsp->frequency != NATIVE_FREQUENCY)
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{
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long* dst[2] = {&resample_buf[0], &resample_buf[RESAMPLE_BUF_SIZE / 2]};
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int32_t* dst[2] = {&resample_buf[0], &resample_buf[RESAMPLE_BUF_SIZE / 2]};
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if (dsp->frequency < NATIVE_FREQUENCY)
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{
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@ -460,7 +460,7 @@ static inline int resample(long* src[], int count)
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return new_count;
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}
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static inline long clip_sample(long sample, long min, long max)
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static inline long clip_sample(int32_t sample, int32_t min, int32_t max)
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{
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if (sample > max)
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{
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@ -478,13 +478,13 @@ static inline long clip_sample(long sample, long min, long max)
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* taken from the coolplayer project - coolplayer.sourceforge.net
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*/
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static long dither_sample(long sample, long bias, long mask,
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static long dither_sample(int32_t sample, int32_t bias, int32_t mask,
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struct dither_data* dither)
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{
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long output;
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long random;
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long min;
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long max;
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int32_t output;
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int32_t random;
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int32_t min;
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int32_t max;
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/* Noise shape and bias */
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@ -523,7 +523,7 @@ static const long crossfeed_coefs[6] ICONST_ATTR = {
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LOW, LOW_COMP, HIGH_NEG, HIGH_COMP, ATT, ATT_COMP
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};
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static void apply_crossfeed(long* src[], int count)
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static void apply_crossfeed(int32_t* src[], int count)
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{
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asm volatile (
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"lea.l crossfeed_data, %%a1 \n"
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@ -598,20 +598,20 @@ static void apply_crossfeed(long* src[], int count)
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);
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}
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#else
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static void apply_crossfeed(long* src[], int count)
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static void apply_crossfeed(int32_t* src[], int count)
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{
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long a; /* accumulator */
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int32_t a; /* accumulator */
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long low_left = crossfeed_data.lowpass[0];
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long low_right = crossfeed_data.lowpass[1];
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long high_left = crossfeed_data.highpass[0];
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long high_right = crossfeed_data.highpass[1];
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int32_t low_left = crossfeed_data.lowpass[0];
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int32_t low_right = crossfeed_data.lowpass[1];
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int32_t high_left = crossfeed_data.highpass[0];
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int32_t high_right = crossfeed_data.highpass[1];
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unsigned int index = crossfeed_data.index;
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long left, right;
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int32_t left, right;
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long * delay_l = crossfeed_data.delay[0];
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long * delay_r = crossfeed_data.delay[1];
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int32_t* delay_l = crossfeed_data.delay[0];
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int32_t* delay_r = crossfeed_data.delay[1];
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int i;
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@ -711,7 +711,7 @@ void dsp_eq_update_data(bool enabled, int band)
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}
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/* Apply EQ filters to those bands that have got it switched on. */
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static void eq_process(long **x, unsigned num)
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static void eq_process(int32_t **x, unsigned num)
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{
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int i;
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unsigned int channels = dsp->stereo_mode != STEREO_MONO ? 2 : 1;
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@ -736,19 +736,19 @@ static void eq_process(long **x, unsigned num)
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* the src array if gain was applied.
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* Note that this must be called before the resampler.
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*/
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static void apply_gain(long* _src[], int _count)
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static void apply_gain(int32_t* _src[], int _count)
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{
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struct dsp_config *my_dsp = dsp;
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if (my_dsp->replaygain)
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{
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long** src = _src;
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int32_t** src = _src;
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int count = _count;
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long* s0 = src[0];
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long* s1 = src[1];
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int32_t* s0 = src[0];
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int32_t* s1 = src[1];
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long gain = my_dsp->replaygain;
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long s;
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int32_t s;
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int i;
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long *d;
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int32_t *d;
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if (s0 != s1)
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{
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@ -773,10 +773,10 @@ static void apply_gain(long* _src[], int _count)
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}
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}
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static void write_samples(short* dst, long* src[], int count)
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static void write_samples(short* dst, int32_t* src[], int count)
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{
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long* s0 = src[0];
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long* s1 = src[1];
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int32_t* s0 = src[0];
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int32_t* s1 = src[1];
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int scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
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if (dsp->dither_enabled)
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@ -815,7 +815,7 @@ static void write_samples(short* dst, long* src[], int count)
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*/
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long dsp_process(char* dst, const char* src[], long size)
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{
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long* tmp[2];
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int32_t* tmp[2];
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long written = 0;
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long factor;
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int samples;
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@ -980,7 +980,7 @@ bool dsp_configure(int setting, void *value)
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else
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{
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dsp->frac_bits = (long) value;
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dsp->sample_bytes = sizeof(long);
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dsp->sample_bytes = 4; /* samples are 32 bits */
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dsp->clip_max = (1 << (long)value) - 1;
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dsp->clip_min = -(1 << (long)value);
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}
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@ -17,6 +17,7 @@
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*
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****************************************************************************/
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#include <inttypes.h>
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#include "config.h"
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#include "eq.h"
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@ -214,7 +215,7 @@ void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, long *c)
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}
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#if (!defined(CPU_COLDFIRE) && !defined(CPU_ARM)) || defined(SIMULATOR)
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void eq_filter(long **x, struct eqfilter *f, unsigned num,
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void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
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unsigned channels, unsigned shift)
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{
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unsigned c, i;
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@ -20,6 +20,8 @@
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#ifndef _EQ_H
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#define _EQ_H
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#include <inttypes.h>
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/* These depend on the fixed point formats used by the different filter types
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and need to be changed when they change.
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*/
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@ -27,14 +29,14 @@
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#define EQ_SHELF_SHIFT 8
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struct eqfilter {
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long coefs[5]; /* Order is b0, b1, b2, a1, a2 */
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long history[2][4];
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int32_t coefs[5]; /* Order is b0, b1, b2, a1, a2 */
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int32_t history[2][4];
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};
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void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
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void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
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void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
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void eq_filter(long **x, struct eqfilter *f, unsigned num,
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void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
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unsigned channels, unsigned shift);
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#endif
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