Encoders: Add a little dithering with the fractional bit for mono mixdowns so faster shifts can be used again instead of division without introducing their own DC offset into the mixed channels.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12246 a1c6a512-1295-4272-9138-f99709370657

4 files changed, 34 insertions, 12 deletions

diff --git a/apps/codecs/aiff_enc.c b/apps/codecs/aiff_enc.c
index f1569d20ff..5f993084c7 100644
--- a/apps/codecs/aiff_enc.c
+++ b/apps/codecs/aiff_enc.c
@@ -69,9 +69,10 @@ struct aiff_header aiff_header =
 
 /* (*) updated when finalizing file */
 
-static int    num_channels;
-uint32_t      sample_rate;
-uint32_t      enc_size;
+static int      num_channels IBSS_ATTR;
+static uint32_t sample_rate;
+static uint32_t enc_size;
+static int32_t  err          IBSS_ATTR;
 
 /* convert unsigned 32 bit value to 80-bit floating point number */
 static void uint32_h_to_ieee754_extended_be(uint8_t f[10], uint32_t l) ICODE_ATTR;
@@ -240,10 +241,14 @@ static void chunk_to_aiff_format(uint32_t *src, uint32_t *dst)
             int32_t lr1, lr2;
 
             lr1 = *(*src)++;
-            lr1 = ((int16_t)lr1 + (lr1 >> 16)) / 2;
+            lr1 = (int16_t)lr1 + (lr1 >> 16) + err;
+            err = lr1 & 1;
+            lr1 >>= 1;
 
             lr2 = *(*src)++;
-            lr2 = ((int16_t)lr2 + (lr2 >> 16)) / 2;
+            lr2 = (int16_t)lr2 + (lr2 >> 16) + err;
+            err = lr2 & 1;
+            lr2 >>= 1;
             *(*dst)++ = swap_odd_even_le32((lr1 << 16) | (uint16_t)lr2);
         } /* to_mono */
 
@@ -311,6 +316,7 @@ static bool init_encoder(void)
 
     sample_rate  = inputs.sample_rate;
     num_channels = inputs.num_channels;
+    err          = 0;
 
     /* configure the buffer system */
     params.afmt            = AFMT_AIFF;
diff --git a/apps/codecs/mp3_enc.c b/apps/codecs/mp3_enc.c
index 64c0996dc2..70b327aef2 100644
--- a/apps/codecs/mp3_enc.c
+++ b/apps/codecs/mp3_enc.c
@@ -174,6 +174,7 @@ static unsigned samp_per_frame                   IBSS_ATTR;
 
 static config_t          cfg                     IBSS_ATTR;
 static char             *res_buffer;
+static int32_t           err                     IBSS_ATTR;
 
 static const uint8_t ht_count_const[2][2][16] =
 { { { 1,  5,  4,  5,  6,  5, 4, 4, 7, 3, 6, 0, 7, 2, 3, 1 },     /* table0 */
@@ -2055,7 +2056,9 @@ static void to_mono_mm(void)
     inline void to_mono(uint32_t **samp)
     {
         int32_t lr = **samp;
-        int32_t m  = ((int16_t)lr + (lr >> 16)) / 2;
+        int32_t m  = (int16_t)lr + (lr >> 16) + err;
+        err = m & 1;
+        m >>= 1;
         *(*samp)++ = (m << 16) | (uint16_t)m;
     } /* to_mono */
 
@@ -2434,6 +2437,8 @@ static bool enc_init(void)
     init_mp3_encoder_engine(inputs.sample_rate, inputs.num_channels,
                             inputs.config);
 
+    err = 0;
+
     /* configure the buffer system */
     params.afmt            = AFMT_MPA_L3;
     params.chunk_size      = cfg.byte_per_frame + 1;
diff --git a/apps/codecs/wav_enc.c b/apps/codecs/wav_enc.c
index 4b7b0083c8..dddca3f5c2 100644
--- a/apps/codecs/wav_enc.c
+++ b/apps/codecs/wav_enc.c
@@ -57,9 +57,10 @@ struct riff_header
 #define PCM_SAMP_PER_CHUNK       2048
 #define PCM_CHUNK_SIZE          (PCM_SAMP_PER_CHUNK*4)
 
-static int    num_channels;
-uint32_t      sample_rate;
-uint32_t      enc_size;
+static int      num_channels IBSS_ATTR;
+static uint32_t sample_rate;
+static uint32_t enc_size;
+static int32_t  err          IBSS_ATTR;
 
 static const struct riff_header riff_header =
 {
@@ -229,10 +230,14 @@ static void chunk_to_wav_format(uint32_t *src, uint32_t *dst)
             int32_t lr1, lr2;
 
             lr1 = *(*src)++;
-            lr1 = ((int16_t)lr1 + (lr1 >> 16)) / 2;
+            lr1 = (int16_t)lr1 + (lr1 >> 16) + err;
+            err = lr1 & 1;
+            lr1 >>= 1;
 
             lr2 = *(*src)++;
-            lr2 = ((int16_t)lr2 + (lr2 >> 16)) / 2;
+            lr2 = (int16_t)lr2 + (lr2 >> 16) + err;
+            err = lr2 & 1;
+            lr2 >>= 1; 
             *(*dst)++ = swap_odd_even_be32((lr1 << 16) | (uint16_t)lr2);
         } /* to_mono */
 
@@ -300,6 +305,7 @@ static bool init_encoder(void)
 
     sample_rate  = inputs.sample_rate;
     num_channels = inputs.num_channels;
+    err          = 0;
 
     /* configure the buffer system */
     params.afmt            = AFMT_PCM_WAV;
diff --git a/apps/codecs/wavpack_enc.c b/apps/codecs/wavpack_enc.c
index 208cd3bc68..71222597e0 100644
--- a/apps/codecs/wavpack_enc.c
+++ b/apps/codecs/wavpack_enc.c
@@ -69,6 +69,7 @@ static int8_t input_buffer[PCM_CHUNK_SIZE*2]     IBSS_ATTR;
 static WavpackConfig config                      IBSS_ATTR;
 static WavpackContext *wpc;
 static int32_t data_size, input_size, input_step IBSS_ATTR;
+static int32_t err                               IBSS_ATTR;
 
 static const WavpackMetadataHeader wvpk_mdh =
 {
@@ -126,7 +127,9 @@ static void chunk_to_int32(int32_t *src)
         {
             int32_t t = *(*src)++;
             /* endianness irrelevant */
-            *(*dst)++ = ((int16_t)t + (t >> 16)) / 2;
+            t = (int16_t)t + (t >> 16) + err;
+            err = t & 1;
+            *(*dst)++ = t >> 1;
         } /* to_int32 */
 
         do
@@ -364,6 +367,8 @@ static bool init_encoder(void)
     if (!WavpackSetConfiguration(wpc, &config, -1))
         return false;
 
+    err = 0;
+
     /* configure the buffer system */
     params.afmt            = AFMT_WAVPACK;
     input_size             = PCM_CHUNK_SIZE*inputs.num_channels / 2;