len0rd/rockbox
1
0
Fork 1
mirror of https://github.com/Rockbox/rockbox.git synced 2025-10-13 18:17:39 -04:00
Code Issues Projects Releases Packages Wiki Activity Actions

Fixes FS#10572: Timestretch + limiter is broken

Browse source 
git-svn-id: svn://svn.rockbox.org/rockbox/branches/v3_4@22774 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Jeffrey Goode 2009-09-21 19:04:39 +00:00
parent 9cbf1c3886
commit 07d5d7e600
1 changed files with 31 additions and 27 deletions
Download patch file Download diff file Expand all files Collapse all files

58
apps/dsp.c
View file

@ -227,7 +227,37 @@ static long album_peak;
static long replaygain;
static bool crossfeed_enabled;
#define AUDIO_DSP (dsp_conf[CODEC_IDX_AUDIO])
#define VOICE_DSP (dsp_conf[CODEC_IDX_VOICE])
/* The internal format is 32-bit samples, non-interleaved, stereo. This
* format is similar to the raw output from several codecs, so the amount
* of copying needed is minimized for that case.
*/
#define RESAMPLE_RATIO 4 /* Enough for 11,025 Hz -> 44,100 Hz */
static int32_t small_sample_buf[SMALL_SAMPLE_BUF_COUNT] IBSS_ATTR;
static int32_t small_resample_buf[SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO] IBSS_ATTR;
static int32_t *big_sample_buf = NULL;
static int32_t *big_resample_buf = NULL;
static int big_sample_buf_count = -1; /* -1=unknown, 0=not available */
static int sample_buf_count;
static int32_t *sample_buf;
static int32_t *resample_buf;
#define SAMPLE_BUF_LEFT_CHANNEL 0
#define SAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2)
#define RESAMPLE_BUF_LEFT_CHANNEL 0
#define RESAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2 * RESAMPLE_RATIO)
/* limiter */
/* MAX_COUNT is largest possible sample count in limiter_process. This is
needed in case time stretch makes the count in dsp_process larger than
the limiter buffer. */
#define MAX_COUNT MAX(SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO / 2, LIMITER_BUFFER_SIZE)
static int count_adjust;
static bool limiter_buffer_active;
static bool limiter_buffer_full;
@ -238,7 +268,7 @@ static int32_t *start_lim_buf[2] IBSS_ATTR,
static uint16_t lim_buf_peak[LIMITER_BUFFER_SIZE] IBSS_ATTR;
static uint16_t *start_peak IBSS_ATTR,
*end_peak IBSS_ATTR;
static uint16_t out_buf_peak[LIMITER_BUFFER_SIZE] IBSS_ATTR;
static uint16_t out_buf_peak[MAX_COUNT];
static uint16_t *out_buf_peak_index IBSS_ATTR;
static uint16_t release_peak IBSS_ATTR;
static int32_t in_samp IBSS_ATTR,
@ -276,32 +306,6 @@ const long gain_steps[49] ICONST_ATTR = { 0x10000000,
0x4EA84FE, 0x4C6D00E, 0x4A41E78, 0x48268DF, 0x461A81C,
0x441D53E, 0x422E985, 0x404DE62};
#define AUDIO_DSP (dsp_conf[CODEC_IDX_AUDIO])
#define VOICE_DSP (dsp_conf[CODEC_IDX_VOICE])
/* The internal format is 32-bit samples, non-interleaved, stereo. This
* format is similar to the raw output from several codecs, so the amount
* of copying needed is minimized for that case.
*/
#define RESAMPLE_RATIO 4 /* Enough for 11,025 Hz -> 44,100 Hz */
static int32_t small_sample_buf[SMALL_SAMPLE_BUF_COUNT] IBSS_ATTR;
static int32_t small_resample_buf[SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO] IBSS_ATTR;
static int32_t *big_sample_buf = NULL;
static int32_t *big_resample_buf = NULL;
static int big_sample_buf_count = -1; /* -1=unknown, 0=not available */
static int sample_buf_count;
static int32_t *sample_buf;
static int32_t *resample_buf;
#define SAMPLE_BUF_LEFT_CHANNEL 0
#define SAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2)
#define RESAMPLE_BUF_LEFT_CHANNEL 0
#define RESAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2 * RESAMPLE_RATIO)
/* Clip sample to signed 16 bit range */
static inline int32_t clip_sample_16(int32_t sample)