Commit 3a2ba96b authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont
Browse files

aout: rewrite and robustify conversion pipelines

This should fix problems when remixing is required but the FL32
format is not involved, as well as decoding on non-FPU platforms.

This also disables (or rather avoids) remixing in A52 and DTS decoders.
I do not really see the point in using the A52 downmixer anyway.
parent 66ede74c
......@@ -82,50 +82,6 @@ static filter_t * FindFilter( vlc_object_t *obj,
return p_filter;
}
/**
* Splits audio format conversion in two simpler conversions
* @return 0 on successful split, -1 if the input and output formats are too
* similar to split the conversion.
*/
static int SplitConversion( const audio_sample_format_t *restrict infmt,
const audio_sample_format_t *restrict outfmt,
audio_sample_format_t *midfmt )
{
*midfmt = *outfmt;
/* Lastly: resample (after format conversion and remixing) */
if( infmt->i_rate != outfmt->i_rate )
midfmt->i_rate = infmt->i_rate;
else
/* Penultimately: remix channels (after format conversion) */
if( infmt->i_physical_channels != outfmt->i_physical_channels
|| infmt->i_original_channels != outfmt->i_original_channels )
{
midfmt->i_physical_channels = infmt->i_physical_channels;
midfmt->i_original_channels = infmt->i_original_channels;
}
else
/* Second: convert linear to S16N as intermediate format */
if( AOUT_FMT_LINEAR( infmt ) )
{
/* All conversion from linear to S16N must be supported directly. */
if( outfmt->i_format == VLC_CODEC_S16N )
return -1;
midfmt->i_format = VLC_CODEC_S16N;
}
else
/* First: convert non-linear to FI32 as intermediate format */
{
if( outfmt->i_format == VLC_CODEC_FI32 )
return -1;
midfmt->i_format = VLC_CODEC_FI32;
}
assert( !AOUT_FMTS_IDENTICAL( infmt, midfmt ) );
aout_FormatPrepare( midfmt );
return 0;
}
/**
* Destroys a chain of audio filters.
*/
......@@ -140,77 +96,175 @@ static void aout_FiltersPipelineDestroy(filter_t *const *filters, unsigned n)
}
}
static filter_t *TryFormat (vlc_object_t *obj, vlc_fourcc_t codec,
audio_sample_format_t *restrict fmt)
{
audio_sample_format_t output = *fmt;
assert (codec != fmt->i_format);
output.i_format = codec;
aout_FormatPrepare (&output);
filter_t *filter = FindFilter (obj, fmt, &output);
if (filter != NULL)
*fmt = output;
return filter;
}
/**
* Allocates audio format conversion filters
* @param obj parent VLC object for new filters
* @param filters table of filters [IN/OUT]
* @param nb_filters pointer to the number of filters in the table [IN/OUT]
* @param max_filters size of filters table [IN]
* @param count pointer to the number of filters in the table [IN/OUT]
* @param max size of filters table [IN]
* @param infmt input audio format
* @param outfmt output audio format
* @return 0 on success, -1 on failure
*/
static int aout_FiltersPipelineCreate(vlc_object_t *obj, filter_t **filters,
unsigned *nb_filters, unsigned max_filters,
unsigned *count, unsigned max,
const audio_sample_format_t *restrict infmt,
const audio_sample_format_t *restrict outfmt)
{
audio_sample_format_t curfmt = *outfmt;
unsigned i = 0;
max_filters -= *nb_filters;
filters += *nb_filters;
aout_FormatsPrint( obj, "filter(s)", infmt, outfmt );
aout_FormatsPrint (obj, "conversion:", infmt, outfmt);
max -= *count;
filters += *count;
/* There is a lot of second guessing on what the conversion plugins can
* and cannot do. This seems hardly avoidable, the conversion problem need
* to be reduced somehow. */
audio_sample_format_t input = *infmt;
bool same_codec = infmt->i_format == outfmt->i_format;
bool same_rate = infmt->i_rate == outfmt->i_rate;
bool same_mix = infmt->i_physical_channels == outfmt->i_physical_channels
&& infmt->i_original_channels == outfmt->i_original_channels;
unsigned n = 0;
while( !AOUT_FMTS_IDENTICAL( infmt, &curfmt ) )
/* Encapsulate or decode non-linear formats */
if (!AOUT_FMT_LINEAR(infmt) && !same_codec)
{
if( i >= max_filters )
if (n == max)
goto overflow;
filter_t *f = NULL;
if (!AOUT_FMT_LINEAR(outfmt))
f = TryFormat (obj, outfmt->i_format, &input);
if (f == NULL)
f = TryFormat (obj, VLC_CODEC_FI32, &input);
if (f == NULL)
f = TryFormat (obj, VLC_CODEC_FL32, &input);
if (f == NULL)
{
msg_Err( obj, "maximum of %u filters reached", max_filters );
dialog_Fatal( obj, _("Audio filtering failed"),
_("The maximum number of filters (%u) was reached."),
max_filters );
goto rollback;
msg_Err (obj, "cannot find %s for conversion pipeline",
"decoder");
goto error;
}
/* Make room and prepend a filter */
memmove( filters + 1, filters, i * sizeof( *filters ) );
filters[n++] = f;
same_codec = input.i_format == outfmt->i_format;
}
assert (AOUT_FMT_LINEAR(&input));
/* Conversion cannot be done in foreign endianess. */
/* TODO: convert to native endian if needed */
*filters = FindFilter( obj, infmt, &curfmt );
if( *filters != NULL )
/* Remix channels */
if (!same_mix)
{ /* Remixing currently requires FL32... TODO: S16N */
if (input.i_format != VLC_CODEC_FL32)
{
i++;
break; /* done! */
if (n == max)
goto overflow;
filter_t *f = TryFormat (obj, VLC_CODEC_FL32, &input);
if (f == NULL)
{
msg_Err (obj, "cannot find %s for conversion pipeline",
"pre-mix converter");
goto error;
}
filters[n++] = f;
same_codec = input.i_format == outfmt->i_format;
}
audio_sample_format_t midfmt;
/* Split the conversion */
if( SplitConversion( infmt, &curfmt, &midfmt ) )
if (n == max)
goto overflow;
audio_sample_format_t output;
output.i_format = input.i_format;
output.i_rate = input.i_rate;
output.i_physical_channels = outfmt->i_physical_channels;
output.i_original_channels = outfmt->i_original_channels;
aout_FormatPrepare (&output);
filter_t *f = FindFilter (obj, &input, &output);
if (f == NULL)
{
msg_Err( obj, "conversion pipeline failed: %4.4s -> %4.4s",
(const char *)&infmt->i_format,
(const char *)&outfmt->i_format );
goto rollback;
msg_Err (obj, "cannot find %s for conversion pipeline",
"remixer");
goto error;
}
*filters = FindFilter( obj, &midfmt, &curfmt );
if( *filters == NULL )
input = output;
filters[n++] = f;
//same_mix = true;
}
/* Resample */
if (!same_rate)
{ /* Resampling works with any linear format, but may be ugly. */
if (n == max)
goto overflow;
audio_sample_format_t output = input;
output.i_rate = outfmt->i_rate;
filter_t *f = FindFilter (obj, &input, &output);
if (f == NULL)
{
msg_Err( obj, "cannot find filter for simple conversion" );
goto rollback;
msg_Err (obj, "cannot find %s for conversion pipeline",
"resampler");
goto error;
}
curfmt = midfmt;
i++;
input = output;
filters[n++] = f;
//same_rate = true;
}
if (!same_codec)
{
if (max == 0)
goto overflow;
filter_t *f = TryFormat (obj, outfmt->i_format, &input);
if (f == NULL)
{
msg_Err (obj, "cannot find %s for conversion pipeline",
"post-mix converter");
goto error;
}
filters[n++] = f;
//same_codec = true;
}
msg_Dbg( obj, "conversion pipeline completed" );
*nb_filters += i;
/* TODO: convert to foreign endian if needed */
msg_Dbg (obj, "conversion pipeline complete");
*count += n;
return 0;
rollback:
aout_FiltersPipelineDestroy (filters, i);
overflow:
msg_Err (obj, "maximum of %u conversion filters reached", max);
dialog_Fatal (obj, _("Audio filtering failed"),
_("The maximum number of filters (%u) was reached."), max);
error:
aout_FiltersPipelineDestroy (filters, n);
return -1;
}
#define aout_FiltersPipelineCreate(obj,f,n,m,i,o) \
aout_FiltersPipelineCreate(VLC_OBJECT(obj),f,n,m,i,o)
......
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