pcm/PcmConvert: add AudioFormat parameters

Don't use src_format.  In the middle of Convert(), the current
AudioFormat has already been modified, it's now something in between
src_format and dest_format.  This simplifies keeping track of what
remains to be done.
This commit is contained in:
Max Kellermann 2013-11-30 13:00:41 +01:00
parent 3c0c939689
commit 3a666702af
2 changed files with 53 additions and 53 deletions

View File

@ -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:

View File

@ -52,13 +52,6 @@ class PcmConvert {
AudioFormat src_format, dest_format;
/**
* Do we get DSD source data? Then this flag is true and
* src_format.format is set to SampleFormat::FLOAT, because
* the #PcmDsd class will convert it to floating point.
*/
bool is_dsd;
public:
PcmConvert();
~PcmConvert();
@ -92,10 +85,18 @@ public:
Error &error);
private:
ConstBuffer<int16_t> Convert16(ConstBuffer<void> src, Error &error);
ConstBuffer<int32_t> Convert24(ConstBuffer<void> src, Error &error);
ConstBuffer<int32_t> Convert32(ConstBuffer<void> src, Error &error);
ConstBuffer<float> ConvertFloat(ConstBuffer<void> src, Error &error);
ConstBuffer<int16_t> Convert16(ConstBuffer<void> src,
AudioFormat format,
Error &error);
ConstBuffer<int32_t> Convert24(ConstBuffer<void> src,
AudioFormat format,
Error &error);
ConstBuffer<int32_t> Convert32(ConstBuffer<void> src,
AudioFormat format,
Error &error);
ConstBuffer<float> ConvertFloat(ConstBuffer<void> src,
AudioFormat format,
Error &error);
};
extern const Domain pcm_convert_domain;