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Commit dbbb4fad authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont
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aout: drop support for U16B, U16L and S16I

parent a944af6d
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Showing with 14 additions and 113 deletions
modules/audio_filter/channel_mixer/remap.c
View file @ dbbb4fad
......@@ -215,7 +215,6 @@ static void RemapAdd##name( filter_t *p_filter, \
}
DEFINE_REMAP( S16N, int16_t )
DEFINE_REMAP( U16N, uint16_t )
DEFINE_REMAP( S32N, int32_t )
DEFINE_REMAP( FL32, float )
DEFINE_REMAP( FL64, double )
......@@ -230,8 +229,6 @@ static inline remap_fun_t GetRemapFun( audio_format_t *p_format, bool b_add )
{
case VLC_CODEC_S16N:
return RemapAddS16N;
case VLC_CODEC_U16N:
return RemapAddU16N;
case VLC_CODEC_S32N:
return RemapAddS32N;
case VLC_CODEC_FL32:
......@@ -246,8 +243,6 @@ static inline remap_fun_t GetRemapFun( audio_format_t *p_format, bool b_add )
{
case VLC_CODEC_S16N:
return RemapCopyS16N;
case VLC_CODEC_U16N:
return RemapCopyU16N;
case VLC_CODEC_S32N:
return RemapCopyS32N;
case VLC_CODEC_FL32:
......
modules/audio_filter/converter/endian.c
View file @ dbbb4fad
......@@ -44,17 +44,6 @@ vlc_module_begin()
set_callbacks(Open, NULL)
vlc_module_end()
static block_t *Filter16(filter_t *filter, block_t *block)
{
uint16_t *data = (uint16_t *)block->p_buffer;
for (size_t i = 0; i < block->i_buffer / 2; i++)
data[i] = bswap16 (data[i]);
(void) filter;
return block;
}
static block_t *Filter24(filter_t *filter, block_t *block)
{
uint8_t *data = (uint8_t *)block->p_buffer;
......@@ -94,10 +83,8 @@ static block_t *Filter64(filter_t *filter, block_t *block)
static const vlc_fourcc_t list[][2] = {
{ VLC_CODEC_F64B, VLC_CODEC_F64L },
{ VLC_CODEC_F32B, VLC_CODEC_F32L },
{ VLC_CODEC_S16B, VLC_CODEC_S16L },
{ VLC_CODEC_S24B, VLC_CODEC_S24L },
{ VLC_CODEC_S32B, VLC_CODEC_S32L },
{ VLC_CODEC_S16B, VLC_CODEC_S16L },
{ VLC_CODEC_S24B, VLC_CODEC_S24L },
{ VLC_CODEC_S32B, VLC_CODEC_S32L },
};
......@@ -128,9 +115,6 @@ static int Open(vlc_object_t *object)
ok:
switch (src->i_bitspersample) {
case 16:
filter->pf_audio_filter = Filter16;
break;
case 24:
filter->pf_audio_filter = Filter24;
break;
......
modules/audio_filter/converter/format.c
View file @ dbbb4fad
......@@ -190,40 +190,6 @@ static block_t *S16toU8(filter_t *filter, block_t *b)
return b;
}
static block_t *U16toU8(filter_t *filter, block_t *b)
{
VLC_UNUSED(filter);
uint16_t *src = (uint16_t *)b->p_buffer;
uint8_t *dst = (uint8_t *)src;
for (int i = b->i_buffer / 2; i--;)
*dst++ = (*src++) >> 8;
b->i_buffer /= 2;
return b;
}
static block_t *S16toU16(filter_t *filter, block_t *b)
{
VLC_UNUSED(filter);
int16_t *src = (int16_t *)b->p_buffer;
uint16_t *dst = (uint16_t *)src;
for (int i = b->i_buffer / 2; i--;)
*dst++ = (*src++) + 32768;
return b;
}
static block_t *U16toS16(filter_t *filter, block_t *b)
{
VLC_UNUSED(filter);
uint16_t *src = (uint16_t *)b->p_buffer;
int16_t *dst = (int16_t *)src;
for (int i = b->i_buffer / 2; i--;)
*dst++ = (int)(*src++) - 32768;
return b;
}
static block_t *S24toS16(filter_t *filter, block_t *b)
{
VLC_UNUSED(filter);
......@@ -339,14 +305,6 @@ static block_t *Fi32toS16(filter_t *filter, block_t *b)
}
/* */
static void X8toX16(block_t *bdst, const block_t *bsrc)
{
uint8_t *src = (uint8_t *)bsrc->p_buffer;
uint16_t *dst = (uint16_t *)bdst->p_buffer;
for (int i = bsrc->i_buffer; i--;)
*dst++ = (*src++) << 8;
}
static void U8toS16(block_t *bdst, const block_t *bsrc)
{
uint8_t *src = (uint8_t *)bsrc->p_buffer;
......@@ -428,10 +386,6 @@ static const struct {
{ VLC_CODEC_FL32, VLC_CODEC_S16N, Fl32toS16 },
{ VLC_CODEC_S16N, VLC_CODEC_U8, S16toU8 },
{ VLC_CODEC_S16N, VLC_CODEC_U16N, S16toU16 },
{ VLC_CODEC_U16N, VLC_CODEC_U8, U16toU8 },
{ VLC_CODEC_U16N, VLC_CODEC_S16N, U16toS16 },
{ 0, 0, NULL }
};
......@@ -447,7 +401,6 @@ static const struct {
{ VLC_CODEC_S16N, VLC_CODEC_S32N, S16toS32 },
{ VLC_CODEC_S16N, VLC_CODEC_FL32, S16toFl32 },
{ VLC_CODEC_U8, VLC_CODEC_U16N, X8toX16 },
{ VLC_CODEC_U8, VLC_CODEC_S16N, U8toS16 },
{ 0, 0, NULL }
};
......
modules/audio_output/alsa.c
View file @ dbbb4fad
......@@ -336,17 +336,8 @@ static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
case VLC_CODEC_U24L:
pcm_format = SND_PCM_FORMAT_U24_3LE;
break;
case VLC_CODEC_S16B:
pcm_format = SND_PCM_FORMAT_S16_BE;
break;
case VLC_CODEC_S16L:
pcm_format = SND_PCM_FORMAT_S16_LE;
break;
case VLC_CODEC_U16B:
pcm_format = SND_PCM_FORMAT_U16_BE;
break;
case VLC_CODEC_U16L:
pcm_format = SND_PCM_FORMAT_U16_LE;
case VLC_CODEC_S16N:
pcm_format = SND_PCM_FORMAT_S16;
break;
case VLC_CODEC_U8:
pcm_format = SND_PCM_FORMAT_U8;
......
modules/audio_output/file.c
View file @ dbbb4fad
......@@ -87,17 +87,9 @@ static void Flush ( audio_output_t *, bool );
#define WAV_LONGTEXT N_("Instead of writing a raw file, you can add a WAV " \
"header to the file.")
static const char *const format_list[] = { "u8", "u16", "s16", "u16_le",
"s16_le", "u16_be", "s16_be",
"float32", "spdif" };
static const int format_int[] = { VLC_CODEC_U8,
VLC_CODEC_U16N, VLC_CODEC_S16N,
VLC_CODEC_U16L,
VLC_CODEC_S16L,
VLC_CODEC_U16B,
VLC_CODEC_S16B,
VLC_CODEC_F32L,
VLC_CODEC_SPDIFL };
static const char *const format_list[] = { "u8", "s16", "float32", "spdif" };
static const int format_int[] = { VLC_CODEC_U8, VLC_CODEC_S16N,
VLC_CODEC_F32L, VLC_CODEC_SPDIFL };
#define FILE_TEXT N_("Output file")
#define FILE_LONGTEXT N_("File to which the audio samples will be written to. (\"-\" for stdout")
......@@ -217,7 +209,6 @@ static int Start( audio_output_t *p_aout, audio_sample_format_t *restrict fmt )
wh->Format = WAVE_FORMAT_PCM;
wh->BitsPerSample = 8;
break;
case VLC_CODEC_S16L:
default:
wh->Format = WAVE_FORMAT_PCM;
wh->BitsPerSample = 16;
......
modules/audio_output/oss.c
View file @ dbbb4fad
......@@ -124,11 +124,8 @@ static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
case VLC_CODEC_S32L:
format = AFMT_S32_LE;
break;
case VLC_CODEC_S16B:
format = AFMT_S16_BE;
break;
case VLC_CODEC_S16L:
format = AFMT_S16_LE;
case VLC_CODEC_S16N:
format = AFMT_S16;
break;
case VLC_CODEC_U8:
format = AFMT_U8;
......@@ -154,9 +151,8 @@ static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
switch (format)
{
case AFMT_U8: fmt->i_format = VLC_CODEC_U8; break;
case AFMT_S16_BE: fmt->i_format = VLC_CODEC_S16B; break;
case AFMT_S16_LE: fmt->i_format = VLC_CODEC_S16L; break;
case AFMT_U8: fmt->i_format = VLC_CODEC_U8; break;
case AFMT_S16: fmt->i_format = VLC_CODEC_S16N; break;
//case AFMT_S24_BE:
//case AFMT_S24_LE:
case AFMT_S32_BE: fmt->i_format = VLC_CODEC_S32B; break;
......
modules/audio_output/pulse.c
View file @ dbbb4fad
......@@ -676,11 +676,8 @@ static int Start(audio_output_t *aout, audio_sample_format_t *restrict fmt)
case VLC_CODEC_S24L:
ss.format = PA_SAMPLE_S24LE;
break;
case VLC_CODEC_S16B:
ss.format = PA_SAMPLE_S16BE;
break;
case VLC_CODEC_S16L:
ss.format = PA_SAMPLE_S16LE;
case VLC_CODEC_S16N:
ss.format = PA_SAMPLE_S16NE;
break;
case VLC_CODEC_U8:
ss.format = PA_SAMPLE_U8;
......
modules/audio_output/sndio.c
View file @ dbbb4fad
......@@ -122,19 +122,13 @@ static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
fmt->i_format = VLC_CODEC_U8;
break;
case 16:
fmt->i_format = par.sig
? (par.le ? VLC_CODEC_S16L : VLC_CODEC_S16B)
: (par.le ? VLC_CODEC_U16L : VLC_CODEC_U16B);
fmt->i_format = VLC_CODEC_S16N;
break;
case 24:
fmt->i_format = par.sig
? (par.le ? VLC_CODEC_S24L : VLC_CODEC_S24B)
: (par.le ? VLC_CODEC_U24L : VLC_CODEC_U24B);
fmt->i_format = VLC_CODEC_S24N;
break;
case 32:
fmt->i_format = par.sig
? (par.le ? VLC_CODEC_S32L : VLC_CODEC_S32B)
: (par.le ? VLC_CODEC_U32L : VLC_CODEC_U32B);
fmt->i_format = VLC_CODEC_S32N;
break;
default:
msg_Err (obj, "unsupported audio sample format (%u bits)",
......
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