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.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@28973 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 35 insertions, 15 deletions

diff --git a/apps/plugins/mp3_encoder.c b/apps/plugins/mp3_encoder.c
index f36a25241a..3cf2b642f0 100644
--- a/apps/plugins/mp3_encoder.c
+++ b/apps/plugins/mp3_encoder.c
@@ -802,7 +802,19 @@ static const int16_t win[18][4] = {
   {   529, -831,-3747,-2387 },
   {   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);