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-rw-r--r--src/pcm/PcmConvert.cxx44
1 files changed, 18 insertions, 26 deletions
diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx
index 0a3fa6816..d280101b5 100644
--- a/src/pcm/PcmConvert.cxx
+++ b/src/pcm/PcmConvert.cxx
@@ -34,16 +34,12 @@
PcmConvert::PcmConvert()
{
- pcm_resample_init(&resample);
-
pcm_buffer_init(&format_buffer);
pcm_buffer_init(&channels_buffer);
}
PcmConvert::~PcmConvert()
{
- pcm_resample_deinit(&resample);
-
pcm_buffer_deinit(&format_buffer);
pcm_buffer_deinit(&channels_buffer);
}
@@ -52,7 +48,7 @@ void
PcmConvert::Reset()
{
dsd.Reset();
- pcm_resample_reset(&resample);
+ resampler.Reset();
}
inline const int16_t *
@@ -93,11 +89,10 @@ PcmConvert::Convert16(const audio_format *src_format,
}
if (src_format->sample_rate != dest_format->sample_rate) {
- buf = pcm_resample_16(&resample,
- dest_format->channels,
- src_format->sample_rate, buf, len,
- dest_format->sample_rate, &len,
- error_r);
+ buf = resampler.Resample16(dest_format->channels,
+ src_format->sample_rate, buf, len,
+ dest_format->sample_rate, &len,
+ error_r);
if (buf == NULL)
return NULL;
}
@@ -143,11 +138,10 @@ PcmConvert::Convert24(const audio_format *src_format,
}
if (src_format->sample_rate != dest_format->sample_rate) {
- buf = pcm_resample_24(&resample,
- dest_format->channels,
- src_format->sample_rate, buf, len,
- dest_format->sample_rate, &len,
- error_r);
+ buf = resampler.Resample24(dest_format->channels,
+ src_format->sample_rate, buf, len,
+ dest_format->sample_rate, &len,
+ error_r);
if (buf == NULL)
return NULL;
}
@@ -193,11 +187,10 @@ PcmConvert::Convert32(const audio_format *src_format,
}
if (src_format->sample_rate != dest_format->sample_rate) {
- buf = pcm_resample_32(&resample,
- dest_format->channels,
- src_format->sample_rate, buf, len,
- dest_format->sample_rate, &len,
- error_r);
+ buf = resampler.Resample32(dest_format->channels,
+ src_format->sample_rate, buf, len,
+ dest_format->sample_rate, &len,
+ error_r);
if (buf == NULL)
return buf;
}
@@ -250,12 +243,11 @@ PcmConvert::ConvertFloat(const audio_format *src_format,
libsamplerate */
if (src_format->sample_rate != dest_format->sample_rate) {
- buffer = pcm_resample_float(&resample,
- dest_format->channels,
- src_format->sample_rate,
- buffer, size,
- dest_format->sample_rate, &size,
- error_r);
+ buffer = resampler.ResampleFloat(dest_format->channels,
+ src_format->sample_rate,
+ buffer, size,
+ dest_format->sample_rate,
+ &size, error_r);
if (buffer == NULL)
return NULL;
}