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Finally fix FS#10678. Now the mp3 encoder plugin supports mono/stereo and the sampling rates 16/22.05/24/32/44.1/48 kHz.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@28976 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Andree Buschmann 2011-01-06 17:36:52 +00:00
parent f0f9f66319
commit ef70c9f32c
1 changed files with 18 additions and 26 deletions
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42
apps/plugins/mp3_encoder.c
View file

@ -15,7 +15,7 @@
#include "plugin.h" #include "plugin.h"
#define SAMP_PER_FRAME 1152 #define MAX_SAMP_PER_FRAME 1152
#define SAMPL2 576 #define SAMPL2 576
#define SBLIMIT 32 #define SBLIMIT 32
#define HTN 16 #define HTN 16
@ -61,7 +61,7 @@ typedef struct {
int ResvSize; int ResvSize;
int channels; int channels;
int granules; int granules;
int resample; int smpl_per_frm;
uint16_t samplerate; uint16_t samplerate;
} config_t; } config_t;
@ -813,7 +813,7 @@ static char* mp3_enc_err[] = {
/* 6 */ "16 bit per sample required.", /* 6 */ "16 bit per sample required.",
/* 7 */ "<=2 channels required.", /* 7 */ "<=2 channels required.",
/* 8 */ "'data' missing.", /* 8 */ "'data' missing.",
/* 9 */ "32/44.1/48 kHz required." /* 9 */ "Samplerate not supported."
}; };
static const char* wav_filename; static const char* wav_filename;
@ -898,12 +898,10 @@ int wave_open(void)
if(cfg.channels > 2) return -7; /* <=2 channels required */ if(cfg.channels > 2) return -7; /* <=2 channels required */
if(!checkString(wavfile,"data")) return -8; if(!checkString(wavfile,"data")) return -8;
/* FIXME: sample rates != 32/44.1/48 kHz do not encode properly as those /* Sample rates != 16/22.05/24/32/44.1/48 kHz are not supported. */
* need MPEG2 format with different setup of the encoder. This MPEG2 setup if((cfg.samplerate != 16000) && (cfg.samplerate != 22050) &&
* is buggy. */ (cfg.samplerate != 24000) && (cfg.samplerate != 32000) &&
if((cfg.samplerate != 32000) && (cfg.samplerate != 44100) && (cfg.samplerate != 48000)) return -9;
(cfg.samplerate != 44100) &&
(cfg.samplerate != 48000)) return -9;
header_size = 0x28; header_size = 0x28;
wav_size = rb->filesize(wavfile); wav_size = rb->filesize(wavfile);
@ -914,7 +912,7 @@ int wave_open(void)
int read_samples(uint16_t *buffer, int num_samples) int read_samples(uint16_t *buffer, int num_samples)
{ {
uint16_t tmpbuf[SAMP_PER_FRAME*2]; /* SAMP_PER_FRAME*MAX_CHANNELS */ uint16_t tmpbuf[MAX_SAMP_PER_FRAME*2]; /* SAMP_PER_FRAME*MAX_CHANNELS */
int byte_per_sample = cfg.channels * 2; /* requires bits_per_sample==16 */ int byte_per_sample = cfg.channels * 2; /* requires bits_per_sample==16 */
int s, samples = rb->read(wavfile, tmpbuf, byte_per_sample * num_samples) / byte_per_sample; int s, samples = rb->read(wavfile, tmpbuf, byte_per_sample * num_samples) / byte_per_sample;
/* Pad last sample with zeros */ /* Pad last sample with zeros */
@ -2092,13 +2090,13 @@ void init_mp3_encoder_engine(bool stereo, int bitrate, uint16_t sample_rate)
if(0 == cfg.mpg.type) if(0 == cfg.mpg.type)
{ /* use MPEG2 format */ { /* use MPEG2 format */
cfg.resample = 1; cfg.smpl_per_frm = MAX_SAMP_PER_FRAME/2;
cfg.granules = 1; cfg.granules = 1;
} }
else else
{ /* use MPEG1 format */ { /* use MPEG1 format */
cfg.resample = 0; cfg.smpl_per_frm = MAX_SAMP_PER_FRAME;
cfg.granules = 2; cfg.granules = 2;
} }
scalefac = sfBand[cfg.mpg.smpl_id + 3*cfg.mpg.type]; scalefac = sfBand[cfg.mpg.smpl_id + 3*cfg.mpg.type];
@ -2172,7 +2170,8 @@ void compress(void)
{ {
if((frames & 7) == 0) if((frames & 7) == 0)
{ rb->lcd_clear_display(); { rb->lcd_clear_display();
rb->lcd_putsxyf(4, 20, "Frame %d / %d", frames, wav_size/SAMPL2/8); rb->lcd_putsxyf(4, 20, "Frame %d / %d", frames,
wav_size/cfg.smpl_per_frm/cfg.channels/2);
rb->lcd_update(); rb->lcd_update();
} }
/* encode one mp3 frame in this loop */ /* encode one mp3 frame in this loop */
@ -2194,25 +2193,18 @@ void compress(void)
memcpy(mfbuf, mfbuf + 2*cfg.granules*576, 4*512); memcpy(mfbuf, mfbuf + 2*cfg.granules*576, 4*512);
/* read new samples to iram for further processing */ /* read new samples to iram for further processing */
if(read_samples((mfbuf + 2*512), SAMP_PER_FRAME) == 0) if(read_samples((mfbuf + 2*512), cfg.smpl_per_frm) == 0)
break; break;
/* swap bytes if neccessary */ /* swap bytes if neccessary */
if(cfg.byte_order == order_bigEndian) if(cfg.byte_order == order_bigEndian)
for(i=0; i<SAMP_PER_FRAME; i++) for(i=0; i<cfg.smpl_per_frm; i++)
{ {
uint32_t t = ((uint32_t*)mfbuf)[512 + i]; uint32_t t = ((uint32_t*)mfbuf)[512 + i];
t = ((t >> 8) & 0xff00ff) | ((t << 8) & 0xff00ff00); t = ((t >> 8) & 0xff00ff) | ((t << 8) & 0xff00ff00);
((uint32_t*)mfbuf)[512 + i] = t; ((uint32_t*)mfbuf)[512 + i] = t;
} }
if(cfg.resample) /* downsample to half of original */
for(i=2*512; i<2*512+2*SAMP_PER_FRAME; i+=4)
{
mfbuf[i/2+512] = (short)(((int)mfbuf[i+0] + mfbuf[i+2]) >> 1);
mfbuf[i/2+513] = (short)(((int)mfbuf[i+1] + mfbuf[i+3]) >> 1);
}
cfg.ResvSize = 0; cfg.ResvSize = 0;
gr_cnt = cfg.granules * cfg.channels; gr_cnt = cfg.granules * cfg.channels;
CodedData.bitpos = cfg.sideinfo_len; /* leave space for mp3 header */ CodedData.bitpos = cfg.sideinfo_len; /* leave space for mp3 header */
@ -2612,7 +2604,7 @@ enum plugin_status plugin_start(const void* parameter)
rb->lcd_clear_display(); rb->lcd_clear_display();
rb->lcd_putsxyf(0, 30, " Conversion: %ld.%02lds ", tim/100, tim%100); rb->lcd_putsxyf(0, 30, " Conversion: %ld.%02lds ", tim/100, tim%100);
tim = frames * SAMP_PER_FRAME * 100 / cfg.samplerate; /* unit=.01s */ tim = frames * cfg.smpl_per_frm * 100 / cfg.samplerate; /* unit=.01s */
rb->lcd_putsxyf(0, 20, " WAV-Length: %ld.%02lds ", tim/100, tim%100); rb->lcd_putsxyf(0, 20, " WAV-Length: %ld.%02lds ", tim/100, tim%100);
rb->lcd_update(); rb->lcd_update();
rb->sleep(5*HZ); rb->sleep(5*HZ);