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Related to FS#10678. The mp3 encoder plugin of Rockbox does not encode MPEG2 layer3 properly and does not support MPEG2.5 layer3 at all. Therefor only samplerates of 32, 44.1 and 48 kHz are supported as input. This change introduces a proper error handling and user error message until the underlying error is fixed.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@28973 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Andree Buschmann 2011-01-06 12:03:12 +00:00
parent 89a7a8138e
commit b5b41e9f68
1 changed files with 35 additions and 15 deletions
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48
apps/plugins/mp3_encoder.c
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@ -803,6 +803,18 @@ static const int16_t win[18][4] = {
{ 362, -471,-3579,-2747 },
{ 134, -146,-3352,-3072 } };
static char* mp3_enc_err[] = {
/* 0 */ "",
/* 1 */ "Cannot open file.",
/* 2 */ "'RIFF' missing.",
/* 3 */ "'WAVE' missing.",
/* 4 */ "'fmt ' missing.",
/* 5 */ "Linear PCM required.",
/* 6 */ "16 bit per sample required.",
/* 7 */ "<=2 channels required.",
/* 8 */ "'data' missing.",
/* 9 */ "32/44.1/48 kHz required."
};
static const char* wav_filename;
static int mp3file, wavfile, wav_size, frames;
@ -881,11 +893,18 @@ int wave_open(void)
wBlockAlign = Read16BitsLowHigh(wavfile);
bits_per_samp = Read16BitsLowHigh(wavfile);
if(wFormatTag != 0x0001) return -5;
if(bits_per_samp != 16) return -6;
if(cfg.channels > 2) return -7;
if(wFormatTag != 0x0001) return -5; /* linear PCM required */
if(bits_per_samp != 16) return -6; /* 16 bps required */
if(cfg.channels > 2) return -7; /* <=2 channels required */
if(!checkString(wavfile,"data")) return -8;
/* FIXME: sample rates != 32/44.1/48 kHz do not encode properly as those
* need MPEG2 format with different setup of the encoder. This MPEG2 setup
* is buggy. */
if((cfg.samplerate != 32000) &&
(cfg.samplerate != 44100) &&
(cfg.samplerate != 48000)) return -9;
header_size = 0x28;
wav_size = rb->filesize(wavfile);
rb->lseek(wavfile, header_size, SEEK_SET);
@ -2063,9 +2082,16 @@ void init_mp3_encoder_engine(bool stereo, int bitrate, uint16_t sample_rate)
cfg.byte_order = order_bigEndian;
#endif
if(bitrate < 96 && stereo && sample_rate >= 32000)
cfg.samplerate = sample_rate;
cfg.channels = stereo ? 2 : 1;
cfg.mpg.mode = stereo ? 0 : 3; /* 0=stereo, 3=mono */
cfg.mpg.bitrate = stereo ? bitrate : bitrate > 160 ? 160 : bitrate;
cfg.mpg.smpl_id = find_samplerate_index(cfg.samplerate, &cfg.mpg.type);
cfg.mpg.bitr_id = find_bitrate_index(cfg.mpg.type, cfg.mpg.bitrate);
cfg.mpg.num_bands = num_bands[stereo ? cfg.mpg.type : 2][cfg.mpg.bitr_id];
if(0 == cfg.mpg.type)
{ /* use MPEG2 format */
sample_rate >>= 1;
cfg.resample = 1;
cfg.granules = 1;
}
@ -2075,14 +2101,6 @@ void init_mp3_encoder_engine(bool stereo, int bitrate, uint16_t sample_rate)
cfg.granules = 2;
}
cfg.samplerate = sample_rate;
cfg.channels = stereo ? 2 : 1;
cfg.mpg.mode = stereo ? 0 : 3; /* 0=stereo, 3=mono */
cfg.mpg.bitrate = stereo ? bitrate : bitrate > 160 ? 160 : bitrate;
cfg.mpg.smpl_id = find_samplerate_index(cfg.samplerate, &cfg.mpg.type);
cfg.mpg.bitr_id = find_bitrate_index(cfg.mpg.type, cfg.mpg.bitrate);
cfg.mpg.num_bands = num_bands[stereo ? cfg.mpg.type : 2][cfg.mpg.bitr_id];
scalefac = sfBand[cfg.mpg.smpl_id + 3*cfg.mpg.type];
ht[ 0].table = NULL; ht[ 0].hlen = NULL; /* Apparently not used */
@ -2585,14 +2603,16 @@ enum plugin_status plugin_start(const void* parameter)
else
{
rb->close(wavfile);
rb->lcd_clear_display();
rb->lcd_putsxyf(0, 20, "WaveOpen failed %d", ret);
rb->lcd_putsxyf(0, 30, "%s", mp3_enc_err[-ret]);
rb->lcd_update();
rb->sleep(5*HZ);
}
rb->lcd_clear_display();
rb->lcd_putsxyf(0, 30, " Conversion: %ld.%02lds ", tim/100, tim%100);
tim = frames * SAMP_PER_FRAME * 100 / 44100; /* unit=.01s */
tim = frames * SAMP_PER_FRAME * 100 / cfg.samplerate; /* unit=.01s */
rb->lcd_putsxyf(0, 20, " WAV-Length: %ld.%02lds ", tim/100, tim%100);
rb->lcd_update();
rb->sleep(5*HZ);