Add flac + quality + compression level support

Adds flac audio to cli, lingui, and macgui
Adds quality and compression level options to cli
Adds quality option to lingui
Quality option works for vorbis and lame

git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4281 b64f7644-9d1e-0410-96f1-a4d463321fa5

1 files changed, 178 insertions, 5 deletions

diff --git a/libhb/common.c b/libhb/common.c
index 5abc9b950..4620602a7 100644
--- a/libhb/common.c
+++ b/libhb/common.c
@@ -75,6 +75,7 @@ hb_encoder_t hb_audio_encoders[] =
   { "MP3 (lame)",         "lame",       HB_ACODEC_LAME,         HB_MUX_MP4|HB_MUX_MKV },
   { "MP3 Passthru",       "copy:mp3",   HB_ACODEC_MP3_PASS,     HB_MUX_MP4|HB_MUX_MKV },
   { "Vorbis (vorbis)",    "vorbis",     HB_ACODEC_VORBIS,                  HB_MUX_MKV },
+  { "FLAC (ffmpeg)",      "ffflac",     HB_ACODEC_FFFLAC,                  HB_MUX_MKV },
   { "Auto Passthru",      "copy",       HB_ACODEC_AUTO_PASS,    HB_MUX_MP4|HB_MUX_MKV } };
 int hb_audio_encoders_count = sizeof( hb_audio_encoders ) /
                               sizeof( hb_encoder_t );
@@ -211,6 +212,10 @@ int hb_find_closest_audio_bitrate(int bitrate)
     int ii;
     int result;
 
+    // Check if bitrate mode was disabled
+    if( bitrate <= 0 )
+        return bitrate;
+
     // result is highest rate if none found during search.
     // rate returned will always be <= rate asked for.
     result = hb_audio_bitrates[0].rate;
@@ -262,8 +267,19 @@ void hb_get_audio_bitrate_limits(uint32_t codec, int samplerate, int mixdown, in
     int channels;
 
     channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(mixdown);
-    switch (codec)
+    if( codec & HB_ACODEC_PASS_FLAG )
+    {
+        // Bitrates don't apply to "lossless" audio (Passthru, FLAC)
+        *low = *high = -1;
+        return;
+    }
+    switch( codec )
     {
+        case HB_ACODEC_FFFLAC:
+            // Bitrates don't apply to "lossless" audio (Passthru, FLAC)
+            *high = *low = -1;
+            break;
+
         case HB_ACODEC_AC3:
             *low = 32 * channels;
             if (samplerate > 24000)
@@ -432,6 +448,9 @@ int hb_get_default_audio_bitrate( uint32_t codec, int samplerate, int mixdown )
     int bitrate, channels;
     int sr_shift;
 
+    if( codec & HB_ACODEC_PASS_FLAG )
+        return -1;
+
     channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(mixdown);
 
     // Min bitrate is established such that we get good quality
@@ -440,6 +459,10 @@ int hb_get_default_audio_bitrate( uint32_t codec, int samplerate, int mixdown )
 
     switch ( codec )
     {
+        case HB_ACODEC_FFFLAC:
+            bitrate = -1;
+            sr_shift = 0;
+            break;
         case HB_ACODEC_AC3:
             if (channels == 1)
                 bitrate = 96;
@@ -453,12 +476,156 @@ int hb_get_default_audio_bitrate( uint32_t codec, int samplerate, int mixdown )
             break;
         default:
             bitrate = channels * 80;
+            break;
     }
     bitrate >>= sr_shift;
     bitrate = hb_get_best_audio_bitrate( codec, bitrate, samplerate, mixdown );
     return bitrate;
 }
 
+// Get limits and hints for the UIs.
+//
+// granularity sets the minimum step increments that should be used
+// (it's ok to round up to some nice multiple if you like)
+//
+// direction says whether 'low' limit is highest or lowest 
+// quality (direction 0 == lowest value is worst quality)
+void hb_get_audio_quality_limits(uint32_t codec, float *low, float *high, float *granularity, int *direction)
+{
+    switch( codec )
+    {
+        case HB_ACODEC_LAME:
+            *direction = 1;
+            *granularity = 0.5;
+            *low = 0.;
+            *high = 10.0;
+            break;
+
+        case HB_ACODEC_VORBIS:
+            *direction = 0;
+            *granularity = 0.05;
+            *low = 0.;
+            *high = 1.0;
+            break;
+
+        case HB_ACODEC_CA_AAC:
+            *direction = 0;
+            *granularity = 9;
+            *low = 0.;
+            *high = 127.0;
+            break;
+
+        default:
+            *direction = 0;
+            *granularity = 1;
+            *low = *high = -1.;
+            break;
+    }
+}
+
+float hb_get_best_audio_quality( uint32_t codec, float quality)
+{
+    float low, high, granularity;
+    int direction;
+
+    hb_get_audio_quality_limits(codec, &low, &high, &granularity, &direction);
+    if (quality > high)
+        quality = high;
+    if (quality < low)
+        quality = low;
+    return quality;
+}
+
+float hb_get_default_audio_quality( uint32_t codec )
+{
+    float quality;
+    switch( codec )
+    {
+        case HB_ACODEC_LAME:
+            quality = 2.;
+            break;
+
+        case HB_ACODEC_VORBIS:
+            quality = .5;
+            break;
+
+        case HB_ACODEC_CA_AAC:
+            quality = 91.;
+            break;
+
+        default:
+            quality = -1.;
+            break;
+    }
+    return quality;
+}
+
+// Get limits and hints for the UIs.
+//
+// granularity sets the minimum step increments that should be used
+// (it's ok to round up to some nice multiple if you like)
+//
+// direction says whether 'low' limit is highest or lowest 
+// compression level (direction 0 == lowest value is worst compression level)
+void hb_get_audio_compression_limits(uint32_t codec, float *low, float *high, float *granularity, int *direction)
+{
+    switch( codec )
+    {
+        case HB_ACODEC_FFFLAC:
+            *direction = 0;
+            *granularity = 1;
+            *high = 12;
+            *low = 0;
+            break;
+
+        case HB_ACODEC_LAME:
+            *direction = 1;
+            *granularity = 1;
+            *high = 9;
+            *low = 0;
+            break;
+
+        default:
+            *direction = 0;
+            *granularity = 1;
+            *low = *high = -1;
+            break;
+    }
+}
+
+float hb_get_best_audio_compression( uint32_t codec, float compression)
+{
+    float low, high, granularity;
+    int direction;
+
+    hb_get_audio_compression_limits( codec, &low, &high, &granularity, &direction );
+    if( compression > high )
+        compression = high;
+    if( compression < low )
+        compression = low;
+    return compression;
+}
+
+float hb_get_default_audio_compression( uint32_t codec )
+{
+    float compression;
+    switch( codec )
+    {
+        case HB_ACODEC_FFFLAC:
+            compression = 5;
+            break;
+
+        case HB_ACODEC_LAME:
+            compression = 2;
+            break;
+
+        default:
+            compression = -1;
+            break;
+    }
+    return compression;
+}
+
 int hb_get_best_mixdown( uint32_t codec, int layout, int mixdown )
 {
 
@@ -527,7 +694,7 @@ int hb_get_best_mixdown( uint32_t codec, int layout, int mixdown )
     }
     // return the best that is not greater than the requested mixdown
     // 0 means the caller requested the best available mixdown
-    if( best_mixdown > mixdown && mixdown != 0 )
+    if( best_mixdown > mixdown && mixdown > 0 )
         best_mixdown = mixdown;
     
     return best_mixdown;
@@ -539,6 +706,7 @@ int hb_get_default_mixdown( uint32_t codec, int layout )
     switch (codec)
     {
         // the AC3 encoder defaults to the best mixdown up to 6-channel
+        case HB_ACODEC_FFFLAC:
         case HB_ACODEC_AC3:
             mixdown = HB_AMIXDOWN_6CH;
             break;
@@ -1328,11 +1496,14 @@ void hb_audio_config_init(hb_audio_config_t * audiocfg)
     /* Initalize some sensable defaults */
     audiocfg->in.track = audiocfg->out.track = 0;
     audiocfg->out.codec = HB_ACODEC_FAAC;
-    audiocfg->out.bitrate = 128;
-    audiocfg->out.samplerate = 44100;
-    audiocfg->out.mixdown = HB_AMIXDOWN_DOLBYPLII;
+    audiocfg->out.bitrate = -1;
+    audiocfg->out.quality = -1;
+    audiocfg->out.compression_level = -1;
+    audiocfg->out.samplerate = -1;
+    audiocfg->out.mixdown = -1;
     audiocfg->out.dynamic_range_compression = 0;
     audiocfg->out.name = NULL;
+
 }
 
 /**********************************************************************
@@ -1386,6 +1557,8 @@ int hb_audio_add(const hb_job_t * job, const hb_audio_config_t * audiocfg)
         audio->config.out.codec &= ~HB_ACODEC_PASS_FLAG;
         audio->config.out.samplerate = audiocfg->out.samplerate;
         audio->config.out.bitrate = audiocfg->out.bitrate;
+        audio->config.out.compression_level = audiocfg->out.compression_level;
+        audio->config.out.quality = audiocfg->out.quality;
         audio->config.out.dynamic_range_compression = audiocfg->out.dynamic_range_compression;
         audio->config.out.mixdown = audiocfg->out.mixdown;
         audio->config.out.gain = audiocfg->out.gain;