{ "64", 64 }, { "80", 80 }, { "96", 96 }, { "112", 112 },
{ "128", 128 }, { "160", 160 }, { "192", 192 }, { "224", 224 },
{ "256", 256 }, { "320", 320 }, { "384", 384 }, { "448", 448 },
- { "768", 768 } };
+ { "512", 512 }, { "576", 576 }, { "640", 640 }, { "768", 768 } };
int hb_audio_bitrates_count = sizeof( hb_audio_bitrates ) /
sizeof( hb_rate_t );
-int hb_audio_bitrates_default = 8; /* 128 kbps */
static hb_error_handler_t *error_handler = NULL;
return "";
}
+// Given an input bitrate, find closest match in the set of allowed bitrates
+int hb_find_closest_audio_bitrate(int bitrate)
+{
+ int ii;
+ int result;
+
+ // result is highest rate if none found during search.
+ // rate returned will always be <= rate asked for.
+ result = hb_audio_bitrates[0].rate;
+ for (ii = hb_audio_bitrates_count-1; ii >= 0; ii--)
+ {
+ if (bitrate >= hb_audio_bitrates[ii].rate)
+ {
+ result = hb_audio_bitrates[ii].rate;
+ break;
+ }
+ }
+ return result;
+}
+
+// Get the bitrate low and high limits for a codec/samplerate/mixdown triplet
+// The limits have been empirically determined through testing. Max bitrates
+// in table below. Numbers in parenthesis are the target bitrate chosen.
+/*
+Encoder 1 channel 2 channels 6 channels
+
+faac
+24kHz 86 (128) 173 (256) 460 (768)
+48kHz 152 (160) 304 (320) 759 (768)
+
+Vorbis
+24kHz 97 (80) 177 (160) 527 (512)
+48kHz 241 (224) 465 (448) 783 (768)
+
+Lame
+24kHz 146 (768) 138 (768)
+48kHz 318 (768) 318 (768)
+
+ffac3
+24kHz 318 (320) 318 (320) 318 (320)
+48kHz 636 (640) 636 (640) 636 (640)
+
+Core Audio (core audio api provides range of allowed bitrates)
+24kHz 16-64 32-128 80-320
+44.1kHz 64-320 160-768
+48kHz 32-256 64-320 160-768
+
+Core Audio (minimum limits found in testing)
+24kHz 16 32 96
+44.1kHz 32 64 160
+48kHz 40 80 240
+*/
+
+void hb_get_audio_bitrate_limits(uint32_t codec, int samplerate, int mixdown, int *low, int *high)
+{
+ int channels;
+
+ channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(mixdown);
+ switch (codec)
+ {
+ case HB_ACODEC_AC3:
+ *low = 32 * channels;
+ if (samplerate > 24000)
+ {
+ *high = 640;
+ }
+ else
+ {
+ *high = 320;
+ }
+ break;
+
+ case HB_ACODEC_CA_AAC:
+ if (samplerate > 44100)
+ {
+ *low = channels * 40;
+ *high = 256;
+ if (channels == 2)
+ *high = 320;
+ if (channels == 6)
+ {
+ *high = 768;
+ }
+ }
+ else if (samplerate > 24000)
+ {
+ *low = channels * 32;
+ *high = 256;
+ if (channels == 2)
+ *high = 320;
+ if (channels == 6)
+ {
+ *low = 160;
+ *high = 768;
+ }
+ }
+ else
+ {
+ *low = channels * 16;
+ *high = channels * 64;
+ if (channels == 6)
+ {
+ *high = 320;
+ }
+ }
+ break;
+
+ case HB_ACODEC_FAAC:
+ *low = 32 * channels;
+ if (samplerate > 24000)
+ {
+ *high = 160 * channels;
+ if (*high > 768)
+ *high = 768;
+ }
+ else
+ {
+ *high = 96 * channels;
+ if (*high > 480)
+ *high = 480;
+ }
+ break;
+
+ case HB_ACODEC_VORBIS:
+ *high = channels * 80;
+ if (samplerate > 24000)
+ {
+ if (channels > 2)
+ {
+ // Vorbis minimum is around 30kbps/ch for 6ch
+ // at rates > 24k (32k/44.1k/48k)
+ *low = 32 * channels;
+ *high = 128 * channels;
+ }
+ else
+ {
+ // Allow 24kbps mono and 48kbps stereo at rates > 24k
+ // (32k/44.1k/48k)
+ *low = 24 * channels;
+ if (samplerate > 32000)
+ *high = channels * 224;
+ else
+ *high = channels * 160;
+ }
+ }
+ else
+ {
+ *low = channels * 16;
+ *high = 80 * channels;
+ }
+ break;
+
+ case HB_ACODEC_LAME:
+ *low = hb_audio_bitrates[0].rate;
+ if (samplerate > 24000)
+ *high = 320;
+ else
+ *high = 160;
+ break;
+
+ default:
+ *low = hb_audio_bitrates[0].rate;
+ *high = hb_audio_bitrates[hb_audio_bitrates_count-1].rate;
+ break;
+ }
+}
+
+// Given an input bitrate, sanitize it. Check low and high limits and
+// make sure it is in the set of allowed bitrates.
+int hb_get_best_audio_bitrate( uint32_t codec, int bitrate, int samplerate, int mixdown)
+{
+ int low, high;
+
+ hb_get_audio_bitrate_limits(codec, samplerate, mixdown, &low, &high);
+ if (bitrate > high)
+ bitrate = high;
+ if (bitrate < low)
+ bitrate = low;
+ bitrate = hb_find_closest_audio_bitrate(bitrate);
+ return bitrate;
+}
+
+// Get the default bitrate for a given codec/samplerate/mixdown triplet.
+int hb_get_default_audio_bitrate( uint32_t codec, int samplerate, int mixdown )
+{
+ int bitrate, channels;
+ int sr_shift;
+
+ channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(mixdown);
+
+ // Min bitrate is established such that we get good quality
+ // audio as a minimum.
+ sr_shift = (samplerate <= 24000) ? 1 : 0;
+
+ switch ( codec )
+ {
+ case HB_ACODEC_AC3:
+ if (channels == 1)
+ bitrate = 96;
+ else if (channels <= 2)
+ bitrate = 224;
+ else
+ bitrate = 640;
+ break;
+ default:
+ bitrate = channels * 80;
+ }
+ bitrate >>= sr_shift;
+ bitrate = hb_get_best_audio_bitrate( codec, bitrate, samplerate, mixdown );
+ return bitrate;
+}
+
+int hb_get_best_mixdown( uint32_t codec, int layout, int mixdown )
+{
+
+ int best_mixdown;
+
+ if (codec & HB_ACODEC_PASS_FLAG)
+ {
+ // Audio pass-thru. No mixdown.
+ return 0;
+ }
+ switch (layout & HB_INPUT_CH_LAYOUT_DISCRETE_NO_LFE_MASK)
+ {
+ // stereo input or something not handled below
+ default:
+ case HB_INPUT_CH_LAYOUT_STEREO:
+ // mono gets mixed up to stereo & more than stereo gets mixed down
+ best_mixdown = HB_AMIXDOWN_STEREO;
+ break;
+
+ // mono input
+ case HB_INPUT_CH_LAYOUT_MONO:
+ // everything else passes through
+ best_mixdown = HB_AMIXDOWN_MONO;
+ break;
+
+ // dolby (DPL1 aka Dolby Surround = 4.0 matrix-encoded) input
+ // the A52 flags don't allow for a way to distinguish between DPL1 and
+ // DPL2 on a DVD so we always assume a DPL1 source for A52_DOLBY.
+ case HB_INPUT_CH_LAYOUT_DOLBY:
+ best_mixdown = HB_AMIXDOWN_DOLBY;
+ break;
+
+ // 4 channel discrete
+ case HB_INPUT_CH_LAYOUT_2F2R:
+ case HB_INPUT_CH_LAYOUT_3F1R:
+ // a52dec and libdca can't upmix to 6ch,
+ // so we must downmix these.
+ best_mixdown = HB_AMIXDOWN_DOLBYPLII;
+ break;
+
+ // 5, 6, 7, or 8 channel discrete
+ case HB_INPUT_CH_LAYOUT_4F2R:
+ case HB_INPUT_CH_LAYOUT_3F4R:
+ case HB_INPUT_CH_LAYOUT_3F2R:
+ if ( ! ( layout & HB_INPUT_CH_LAYOUT_HAS_LFE ) )
+ {
+ // we don't do 5 channel discrete so mixdown to DPLII
+ // a52dec and libdca can't upmix to 6ch,
+ // so we must downmix this.
+ best_mixdown = HB_AMIXDOWN_DOLBYPLII;
+ }
+ else
+ {
+ switch (codec)
+ {
+ case HB_ACODEC_LAME:
+ best_mixdown = HB_AMIXDOWN_DOLBYPLII;
+ break;
+
+ default:
+ best_mixdown = HB_AMIXDOWN_6CH;
+ break;
+ }
+ }
+ break;
+ }
+ // 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 )
+ best_mixdown = mixdown;
+
+ return best_mixdown;
+}
+
+int hb_get_default_mixdown( uint32_t codec, int layout )
+{
+ int mixdown;
+ switch (codec)
+ {
+ // the AC3 encoder defaults to the best mixdown up to 6-channel
+ case HB_ACODEC_AC3:
+ mixdown = HB_AMIXDOWN_6CH;
+ break;
+ // other encoders default to the best mixdown up to DPLII
+ default:
+ mixdown = HB_AMIXDOWN_DOLBYPLII;
+ break;
+ }
+ // return the best available mixdown up to the selected default
+ return hb_get_best_mixdown( codec, layout, mixdown );
+}
+
/**********************************************************************
* hb_reduce
**********************************************************************
{
hb_title_t * title = job->title;
int i;
+ int min_width;
+ int min_height;
+ int modulus;
/* don't do anything unless the title has complete size info */
if ( title->height == 0 || title->width == 0 || title->aspect == 0 )
return;
}
- /* Sanity checks:
- Widths and heights must be multiples of 16 and greater than or
- equal to 16
- Crop values must be multiples of 2, greater than or equal to 0
- and less than half of the dimension */
- job->width = MULTIPLE_16( job->width );
- job->height = MULTIPLE_16( job->height );
- job->width = MAX( 16, job->width );
- job->height = MAX( 16, job->height );
+ // min_width and min_height should be multiples of modulus
+ min_width = 32;
+ min_height = 32;
+ modulus = job->modulus ? job->modulus : 16;
+
for( i = 0; i < 4; i++ )
{
- job->crop[i] = EVEN( job->crop[i] );
- job->crop[i] = MAX( 0, job->crop[i] );
+ // Sanity check crop values are zero or positive multiples of 2
if( i < 2 )
{
- /* Top, bottom */
- job->crop[i] = MIN( job->crop[i], ( title->height / 2 ) - 2 );
+ // Top, bottom
+ job->crop[i] = MIN( EVEN( job->crop[i] ), EVEN( ( title->height / 2 ) - ( min_height / 2 ) ) );
+ job->crop[i] = MAX( 0, job->crop[i] );
}
else
{
- /* Left, right */
- job->crop[i] = MIN( job->crop[i], ( title->width / 2 ) - 2 );
+ // Left, right
+ job->crop[i] = MIN( EVEN( job->crop[i] ), EVEN( ( title->width / 2 ) - ( min_width / 2 ) ) );
+ job->crop[i] = MAX( 0, job->crop[i] );
}
}
double par = (double)title->width / ( (double)title->height * title->aspect );
double cropped_sar = (double)( title->height - job->crop[0] - job->crop[1] ) /
- (double)(title->width - job->crop[2] - job->crop[3] );
+ (double)( title->width - job->crop[2] - job->crop[3] );
double ar = par * cropped_sar;
+
+ // Dimensions must be greater than minimum and multiple of modulus
if( keep == HB_KEEP_WIDTH )
{
- job->height = MULTIPLE_16( (uint64_t)( (double)job->width * ar ) );
- job->height = MAX( 16, job->height );
+ job->width = MULTIPLE_MOD( job->width, modulus );
+ job->width = MAX( min_width, job->width );
+ job->height = MULTIPLE_MOD( (uint64_t)( (double)job->width * ar ), modulus );
+ job->height = MAX( min_height, job->height );
}
else
{
- job->width = MULTIPLE_16( (uint64_t)( (double)job->height / ar ) );
- job->width = MAX( 16, job->width );
+ job->height = MULTIPLE_MOD( job->height, modulus );
+ job->height = MAX( min_height, job->height );
+ job->width = MULTIPLE_MOD( (uint64_t)( (double)job->height / ar ), modulus );
+ job->width = MAX( min_width, job->width );
}
}
(quite guessed) */
switch( job->mux )
{
- case HB_MUX_MP4:
- case HB_MUX_PSP:
- case HB_MUX_IPOD:
- case HB_MUX_MKV:
- overhead = 6;
- break;
- case HB_MUX_AVI:
- overhead = 24;
- break;
- case HB_MUX_OGM:
+ case HB_MUX_MP4:
+ case HB_MUX_MKV:
overhead = 6;
break;
default:
case HB_ACODEC_LAME:
samples_per_frame = 1152;
break;
+ case HB_ACODEC_AC3_PASS:
+ case HB_ACODEC_DCA_PASS:
case HB_ACODEC_AC3:
case HB_ACODEC_DCA:
samples_per_frame = 1536;
return 0;
}
- if( audio->config.out.codec == HB_ACODEC_AC3 ||
- audio->config.out.codec == HB_ACODEC_DCA)
+ if( audio->config.out.codec == HB_ACODEC_AC3_PASS ||
+ audio->config.out.codec == HB_ACODEC_DCA_PASS)
{
/*
* For pass through we take the bitrate from the input audio
void hb_error( char * log, ... )
{
char string[181]; /* 180 chars + \0 */
+ char rep_string[181];
+ static char last_string[181];
+ static int last_error_count = 0;
+ static uint64_t last_series_error_time = 0;
+ static hb_lock_t *mutex = 0;
va_list args;
+ uint64_t time_now;
/* Convert the message to a string */
va_start( args, log );
vsnprintf( string, 180, log, args );
va_end( args );
+ if( !mutex )
+ {
+ mutex = hb_lock_init();
+ }
+
+ hb_lock( mutex );
+
+ time_now = hb_get_date();
+
+ if( strcmp( string, last_string) == 0 )
+ {
+ /*
+ * The last error and this one are the same, don't log it
+ * just count it instead, unless it was more than one second
+ * ago.
+ */
+ last_error_count++;
+ if( last_series_error_time + ( 1000 * 1 ) > time_now )
+ {
+ hb_unlock( mutex );
+ return;
+ }
+ }
+
+ /*
+ * A new error, or the same one more than 10sec since the last one
+ * did we have any of the same counted up?
+ */
+ if( last_error_count > 0 )
+ {
+ /*
+ * Print out the last error to ensure context for the last
+ * repeated message.
+ */
+ if( error_handler )
+ {
+ error_handler( last_string );
+ } else {
+ hb_log( "%s", last_string );
+ }
+
+ if( last_error_count > 1 )
+ {
+ /*
+ * Only print out the repeat message for more than 2 of the
+ * same, since we just printed out two of them already.
+ */
+ snprintf( rep_string, 180, "Last error repeated %d times",
+ last_error_count - 1 );
+
+ if( error_handler )
+ {
+ error_handler( rep_string );
+ } else {
+ hb_log( "%s", rep_string );
+ }
+ }
+
+ last_error_count = 0;
+ }
+
+ last_series_error_time = time_now;
+
+ strcpy( last_string, string );
+
/*
* Got the error in a single string, send it off to be dispatched.
*/
{
error_handler( string );
} else {
- hb_log( string );
+ hb_log( "%s", string );
}
+
+ hb_unlock( mutex );
}
void hb_register_error_handler( hb_error_handler_t * handler )
**********************************************************************
*
*********************************************************************/
-hb_title_t * hb_title_init( char * dvd, int index )
+hb_title_t * hb_title_init( char * path, int index )
{
hb_title_t * t;
t->list_audio = hb_list_init();
t->list_chapter = hb_list_init();
t->list_subtitle = hb_list_init();
- strcat( t->dvd, dvd );
+ t->list_attachment = hb_list_init();
+ strcat( t->path, path );
// default to decoding mpeg2
t->video_id = 0xE0;
t->video_codec = WORK_DECMPEG2;
hb_audio_t * audio;
hb_chapter_t * chapter;
hb_subtitle_t * subtitle;
+ hb_attachment_t * attachment;
while( ( audio = hb_list_item( t->list_audio, 0 ) ) )
{
while( ( subtitle = hb_list_item( t->list_subtitle, 0 ) ) )
{
hb_list_rem( t->list_subtitle, subtitle );
+ if ( subtitle->extradata )
+ {
+ free( subtitle->extradata );
+ subtitle->extradata = NULL;
+ }
free( subtitle );
}
hb_list_close( &t->list_subtitle );
+
+ while( ( attachment = hb_list_item( t->list_attachment, 0 ) ) )
+ {
+ hb_list_rem( t->list_attachment, attachment );
+ if ( attachment->name )
+ {
+ free( attachment->name );
+ attachment->name = NULL;
+ }
+ if ( attachment->data )
+ {
+ free( attachment->data );
+ attachment->data = NULL;
+ }
+ free( attachment );
+ }
+ hb_list_close( &t->list_attachment );
if( t->metadata )
{
*/
audio->config.out.track = hb_list_count(job->list_audio) + 1;
audio->config.out.codec = audiocfg->out.codec;
- if( audiocfg->out.codec == audio->config.in.codec )
+ if( (audiocfg->out.codec & HB_ACODEC_MASK) == audio->config.in.codec &&
+ (audiocfg->out.codec & HB_ACODEC_PASS_FLAG ) )
{
/* Pass-through, copy from input. */
audio->config.out.samplerate = audio->config.in.samplerate;
else
{
/* Non pass-through, use what is given. */
+ 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.dynamic_range_compression = audiocfg->out.dynamic_range_compression;
{
subtitle = calloc(1, sizeof(*subtitle));
memcpy(subtitle, src, sizeof(*subtitle));
+ if ( src->extradata )
+ {
+ subtitle->extradata = malloc( src->extradata_size );
+ memcpy( subtitle->extradata, src->extradata, src->extradata_size );
+ }
}
return subtitle;
}
}
return retval;
}
+
+char * hb_strdup_printf( char * fmt, ... )
+{
+ int len;
+ va_list ap;
+ int size = 256;
+ char * str;
+ char * tmp;
+
+ str = malloc( size );
+ if ( str == NULL )
+ return NULL;
+
+ while (1)
+ {
+ /* Try to print in the allocated space. */
+ va_start( ap, fmt );
+ len = vsnprintf( str, size, fmt, ap );
+ va_end( ap );
+
+ /* If that worked, return the string. */
+ if ( len > -1 && len < size )
+ {
+ return str;
+ }
+
+ /* Else try again with more space. */
+ if ( len > -1 ) /* glibc 2.1 */
+ size = len + 1; /* precisely what is needed */
+ else /* glibc 2.0 */
+ size *= 2; /* twice the old size */
+ tmp = realloc( str, size );
+ if ( tmp == NULL )
+ {
+ free( str );
+ return NULL;
+ }
+ else
+ str = tmp;
+ }
+}
+
+/**********************************************************************
+ * hb_attachment_copy
+ **********************************************************************
+ *
+ *********************************************************************/
+hb_attachment_t *hb_attachment_copy(const hb_attachment_t *src)
+{
+ hb_attachment_t *attachment = NULL;
+
+ if( src )
+ {
+ attachment = calloc(1, sizeof(*attachment));
+ memcpy(attachment, src, sizeof(*attachment));
+ if ( src->name )
+ {
+ attachment->name = strdup( src->name );
+ }
+ if ( src->data )
+ {
+ attachment->data = malloc( src->size );
+ memcpy( attachment->data, src->data, src->size );
+ }
+ }
+ return attachment;
+}
+
+/**********************************************************************
+ * hb_yuv2rgb
+ **********************************************************************
+ * Converts a YCbCr pixel to an RGB pixel.
+ *
+ * This conversion is lossy (due to rounding and clamping).
+ *
+ * Algorithm:
+ * http://en.wikipedia.org/w/index.php?title=YCbCr&oldid=361987695#Technical_details
+ *********************************************************************/
+int hb_yuv2rgb(int yuv)
+{
+ double y, Cr, Cb;
+ int r, g, b;
+
+ y = (yuv >> 16) & 0xff;
+ Cb = (yuv >> 8) & 0xff;
+ Cr = (yuv ) & 0xff;
+
+ r = 1.164 * (y - 16) + 2.018 * (Cb - 128);
+ g = 1.164 * (y - 16) - 0.813 * (Cr - 128) - 0.391 * (Cb - 128);
+ b = 1.164 * (y - 16) + 1.596 * (Cr - 128);
+
+ r = (r < 0) ? 0 : r;
+ g = (g < 0) ? 0 : g;
+ b = (b < 0) ? 0 : b;
+
+ r = (r > 255) ? 255 : r;
+ g = (g > 255) ? 255 : g;
+ b = (b > 255) ? 255 : b;
+
+ return (r << 16) | (g << 8) | b;
+}
+
+/**********************************************************************
+ * hb_rgb2yuv
+ **********************************************************************
+ * Converts an RGB pixel to a YCbCr pixel.
+ *
+ * This conversion is lossy (due to rounding and clamping).
+ *
+ * Algorithm:
+ * http://en.wikipedia.org/w/index.php?title=YCbCr&oldid=361987695#Technical_details
+ *********************************************************************/
+int hb_rgb2yuv(int rgb)
+{
+ double r, g, b;
+ int y, Cr, Cb;
+
+ r = (rgb >> 16) & 0xff;
+ g = (rgb >> 8) & 0xff;
+ b = (rgb ) & 0xff;
+
+ y = 16. + ( 0.257 * r) + (0.504 * g) + (0.098 * b);
+ Cb = 128. + (-0.148 * r) - (0.291 * g) + (0.439 * b);
+ Cr = 128. + ( 0.439 * r) - (0.368 * g) - (0.071 * b);
+
+ y = (y < 0) ? 0 : y;
+ Cb = (Cb < 0) ? 0 : Cb;
+ Cr = (Cr < 0) ? 0 : Cr;
+
+ y = (y > 255) ? 255 : y;
+ Cb = (Cb > 255) ? 255 : Cb;
+ Cr = (Cr > 255) ? 255 : Cr;
+
+ return (y << 16) | (Cb << 8) | Cr;
+}
+
+const char * hb_subsource_name( int source )
+{
+ switch (source)
+ {
+ case VOBSUB:
+ return "VOBSUB";
+ case SRTSUB:
+ return "SRT";
+ case CC608SUB:
+ return "CC";
+ case CC708SUB:
+ return "CC";
+ case UTF8SUB:
+ return "UTF-8";
+ case TX3GSUB:
+ return "TX3G";
+ case SSASUB:
+ return "SSA";
+ default:
+ return "Unknown";
+ }
+}
+
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