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-rw-r--r--src/pcm/PcmConvert.cxx83
1 files changed, 41 insertions, 42 deletions
diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx
index c405fc94f..e32a4ba4e 100644
--- a/src/pcm/PcmConvert.cxx
+++ b/src/pcm/PcmConvert.cxx
@@ -63,10 +63,6 @@ PcmConvert::Open(AudioFormat _src_format, AudioFormat _dest_format,
src_format = _src_format;
dest_format = _dest_format;
- is_dsd = src_format.format == SampleFormat::DSD;
- if (is_dsd)
- src_format.format = SampleFormat::FLOAT;
-
return true;
}
@@ -83,7 +79,7 @@ PcmConvert::Close()
}
inline ConstBuffer<int16_t>
-PcmConvert::Convert16(ConstBuffer<void> src, Error &error)
+PcmConvert::Convert16(ConstBuffer<void> src, AudioFormat format, Error &error)
{
const int16_t *buf;
size_t len;
@@ -91,34 +87,34 @@ PcmConvert::Convert16(ConstBuffer<void> src, Error &error)
assert(dest_format.format == SampleFormat::S16);
buf = pcm_convert_to_16(format_buffer, dither,
- src_format.format,
+ format.format,
src.data, src.size,
&len);
if (buf == nullptr) {
error.Format(pcm_convert_domain,
"Conversion from %s to 16 bit is not implemented",
- sample_format_to_string(src_format.format));
+ sample_format_to_string(format.format));
return nullptr;
}
- if (src_format.channels != dest_format.channels) {
+ if (format.channels != dest_format.channels) {
buf = pcm_convert_channels_16(channels_buffer,
dest_format.channels,
- src_format.channels,
+ format.channels,
buf, len, &len);
if (buf == nullptr) {
error.Format(pcm_convert_domain,
"Conversion from %u to %u channels "
"is not implemented",
- src_format.channels,
+ format.channels,
dest_format.channels);
return nullptr;
}
}
- if (src_format.sample_rate != dest_format.sample_rate) {
+ if (format.sample_rate != dest_format.sample_rate) {
buf = resampler.Resample16(dest_format.channels,
- src_format.sample_rate, buf, len,
+ format.sample_rate, buf, len,
dest_format.sample_rate, &len,
error);
if (buf == nullptr)
@@ -129,7 +125,7 @@ PcmConvert::Convert16(ConstBuffer<void> src, Error &error)
}
inline ConstBuffer<int32_t>
-PcmConvert::Convert24(ConstBuffer<void> src, Error &error)
+PcmConvert::Convert24(ConstBuffer<void> src, AudioFormat format, Error &error)
{
const int32_t *buf;
size_t len;
@@ -137,33 +133,33 @@ PcmConvert::Convert24(ConstBuffer<void> src, Error &error)
assert(dest_format.format == SampleFormat::S24_P32);
buf = pcm_convert_to_24(format_buffer,
- src_format.format,
+ format.format,
src.data, src.size, &len);
if (buf == nullptr) {
error.Format(pcm_convert_domain,
"Conversion from %s to 24 bit is not implemented",
- sample_format_to_string(src_format.format));
+ sample_format_to_string(format.format));
return nullptr;
}
- if (src_format.channels != dest_format.channels) {
+ if (format.channels != dest_format.channels) {
buf = pcm_convert_channels_24(channels_buffer,
dest_format.channels,
- src_format.channels,
+ format.channels,
buf, len, &len);
if (buf == nullptr) {
error.Format(pcm_convert_domain,
"Conversion from %u to %u channels "
"is not implemented",
- src_format.channels,
+ format.channels,
dest_format.channels);
return nullptr;
}
}
- if (src_format.sample_rate != dest_format.sample_rate) {
+ if (format.sample_rate != dest_format.sample_rate) {
buf = resampler.Resample24(dest_format.channels,
- src_format.sample_rate, buf, len,
+ format.sample_rate, buf, len,
dest_format.sample_rate, &len,
error);
if (buf == nullptr)
@@ -174,7 +170,7 @@ PcmConvert::Convert24(ConstBuffer<void> src, Error &error)
}
inline ConstBuffer<int32_t>
-PcmConvert::Convert32(ConstBuffer<void> src, Error &error)
+PcmConvert::Convert32(ConstBuffer<void> src, AudioFormat format, Error &error)
{
const int32_t *buf;
size_t len;
@@ -182,33 +178,33 @@ PcmConvert::Convert32(ConstBuffer<void> src, Error &error)
assert(dest_format.format == SampleFormat::S32);
buf = pcm_convert_to_32(format_buffer,
- src_format.format,
+ format.format,
src.data, src.size, &len);
if (buf == nullptr) {
error.Format(pcm_convert_domain,
"Conversion from %s to 32 bit is not implemented",
- sample_format_to_string(src_format.format));
+ sample_format_to_string(format.format));
return nullptr;
}
- if (src_format.channels != dest_format.channels) {
+ if (format.channels != dest_format.channels) {
buf = pcm_convert_channels_32(channels_buffer,
dest_format.channels,
- src_format.channels,
+ format.channels,
buf, len, &len);
if (buf == nullptr) {
error.Format(pcm_convert_domain,
"Conversion from %u to %u channels "
"is not implemented",
- src_format.channels,
+ format.channels,
dest_format.channels);
return nullptr;
}
}
- if (src_format.sample_rate != dest_format.sample_rate) {
+ if (format.sample_rate != dest_format.sample_rate) {
buf = resampler.Resample32(dest_format.channels,
- src_format.sample_rate, buf, len,
+ format.sample_rate, buf, len,
dest_format.sample_rate, &len,
error);
if (buf == nullptr)
@@ -219,7 +215,8 @@ PcmConvert::Convert32(ConstBuffer<void> src, Error &error)
}
inline ConstBuffer<float>
-PcmConvert::ConvertFloat(ConstBuffer<void> src, Error &error)
+PcmConvert::ConvertFloat(ConstBuffer<void> src, AudioFormat format,
+ Error &error)
{
assert(dest_format.format == SampleFormat::FLOAT);
@@ -227,27 +224,27 @@ PcmConvert::ConvertFloat(ConstBuffer<void> src, Error &error)
size_t size;
const float *buffer = pcm_convert_to_float(format_buffer,
- src_format.format,
+ format.format,
src.data, src.size, &size);
if (buffer == nullptr) {
error.Format(pcm_convert_domain,
"Conversion from %s to float is not implemented",
- sample_format_to_string(src_format.format));
+ sample_format_to_string(format.format));
return nullptr;
}
/* convert channels */
- if (src_format.channels != dest_format.channels) {
+ if (format.channels != dest_format.channels) {
buffer = pcm_convert_channels_float(channels_buffer,
dest_format.channels,
- src_format.channels,
+ format.channels,
buffer, size, &size);
if (buffer == nullptr) {
error.Format(pcm_convert_domain,
"Conversion from %u to %u channels "
"is not implemented",
- src_format.channels,
+ format.channels,
dest_format.channels);
return nullptr;
}
@@ -256,9 +253,9 @@ PcmConvert::ConvertFloat(ConstBuffer<void> src, Error &error)
/* resample with float, because this is the best format for
libsamplerate */
- if (src_format.sample_rate != dest_format.sample_rate) {
+ if (format.sample_rate != dest_format.sample_rate) {
buffer = resampler.ResampleFloat(dest_format.channels,
- src_format.sample_rate,
+ format.sample_rate,
buffer, size,
dest_format.sample_rate,
&size, error);
@@ -275,10 +272,11 @@ PcmConvert::Convert(const void *src, size_t src_size,
Error &error)
{
ConstBuffer<void> buffer(src, src_size);
+ AudioFormat format = src_format;
- if (is_dsd) {
+ if (format.format == SampleFormat::DSD) {
auto s = ConstBuffer<uint8_t>::FromVoid(buffer);
- auto d = dsd.ToFloat(src_format.channels,
+ auto d = dsd.ToFloat(format.channels,
false, s);
if (d.IsNull()) {
error.Set(pcm_convert_domain,
@@ -287,23 +285,24 @@ PcmConvert::Convert(const void *src, size_t src_size,
}
buffer = d.ToVoid();
+ format.format = SampleFormat::FLOAT;
}
switch (dest_format.format) {
case SampleFormat::S16:
- buffer = Convert16(buffer, error).ToVoid();
+ buffer = Convert16(buffer, format, error).ToVoid();
break;
case SampleFormat::S24_P32:
- buffer = Convert24(buffer, error).ToVoid();
+ buffer = Convert24(buffer, format, error).ToVoid();
break;
case SampleFormat::S32:
- buffer = Convert32(buffer, error).ToVoid();
+ buffer = Convert32(buffer, format, error).ToVoid();
break;
case SampleFormat::FLOAT:
- buffer = ConvertFloat(buffer, error).ToVoid();
+ buffer = ConvertFloat(buffer, format, error).ToVoid();
break;
default:
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-05 02:53:18 +0000