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-rw-r--r--src/pcm/PcmConvert.cxx310
1 files changed, 310 insertions, 0 deletions
diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx
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+++ b/src/pcm/PcmConvert.cxx
@@ -0,0 +1,310 @@
+/*
+ * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "config.h"
+#include "PcmConvert.hxx"
+#include "PcmChannels.hxx"
+#include "PcmFormat.hxx"
+#include "pcm_pack.h"
+#include "AudioFormat.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+
+#include <glib.h>
+
+#include <assert.h>
+#include <math.h>
+
+const Domain pcm_convert_domain("pcm_convert");
+
+PcmConvert::PcmConvert()
+{
+}
+
+PcmConvert::~PcmConvert()
+{
+}
+
+void
+PcmConvert::Reset()
+{
+ dsd.Reset();
+ resampler.Reset();
+}
+
+inline const int16_t *
+PcmConvert::Convert16(const AudioFormat src_format,
+ const void *src_buffer, size_t src_size,
+ const AudioFormat dest_format, size_t *dest_size_r,
+ Error &error)
+{
+ const int16_t *buf;
+ size_t len;
+
+ assert(dest_format.format == SampleFormat::S16);
+
+ buf = pcm_convert_to_16(format_buffer, dither,
+ src_format.format,
+ src_buffer, src_size,
+ &len);
+ if (buf == NULL) {
+ error.Format(pcm_convert_domain,
+ "Conversion from %s to 16 bit is not implemented",
+ sample_format_to_string(src_format.format));
+ return NULL;
+ }
+
+ if (src_format.channels != dest_format.channels) {
+ buf = pcm_convert_channels_16(channels_buffer,
+ dest_format.channels,
+ src_format.channels,
+ buf, len, &len);
+ if (buf == NULL) {
+ error.Format(pcm_convert_domain,
+ "Conversion from %u to %u channels "
+ "is not implemented",
+ src_format.channels,
+ dest_format.channels);
+ return NULL;
+ }
+ }
+
+ if (src_format.sample_rate != dest_format.sample_rate) {
+ buf = resampler.Resample16(dest_format.channels,
+ src_format.sample_rate, buf, len,
+ dest_format.sample_rate, &len,
+ error);
+ if (buf == NULL)
+ return NULL;
+ }
+
+ *dest_size_r = len;
+ return buf;
+}
+
+inline const int32_t *
+PcmConvert::Convert24(const AudioFormat src_format,
+ const void *src_buffer, size_t src_size,
+ const AudioFormat dest_format, size_t *dest_size_r,
+ Error &error)
+{
+ const int32_t *buf;
+ size_t len;
+
+ assert(dest_format.format == SampleFormat::S24_P32);
+
+ buf = pcm_convert_to_24(format_buffer,
+ src_format.format,
+ src_buffer, src_size, &len);
+ if (buf == NULL) {
+ error.Format(pcm_convert_domain,
+ "Conversion from %s to 24 bit is not implemented",
+ sample_format_to_string(src_format.format));
+ return NULL;
+ }
+
+ if (src_format.channels != dest_format.channels) {
+ buf = pcm_convert_channels_24(channels_buffer,
+ dest_format.channels,
+ src_format.channels,
+ buf, len, &len);
+ if (buf == NULL) {
+ error.Format(pcm_convert_domain,
+ "Conversion from %u to %u channels "
+ "is not implemented",
+ src_format.channels,
+ dest_format.channels);
+ return NULL;
+ }
+ }
+
+ if (src_format.sample_rate != dest_format.sample_rate) {
+ buf = resampler.Resample24(dest_format.channels,
+ src_format.sample_rate, buf, len,
+ dest_format.sample_rate, &len,
+ error);
+ if (buf == NULL)
+ return NULL;
+ }
+
+ *dest_size_r = len;
+ return buf;
+}
+
+inline const int32_t *
+PcmConvert::Convert32(const AudioFormat src_format,
+ const void *src_buffer, size_t src_size,
+ const AudioFormat dest_format, size_t *dest_size_r,
+ Error &error)
+{
+ const int32_t *buf;
+ size_t len;
+
+ assert(dest_format.format == SampleFormat::S32);
+
+ buf = pcm_convert_to_32(format_buffer,
+ src_format.format,
+ src_buffer, src_size, &len);
+ if (buf == NULL) {
+ error.Format(pcm_convert_domain,
+ "Conversion from %s to 32 bit is not implemented",
+ sample_format_to_string(src_format.format));
+ return NULL;
+ }
+
+ if (src_format.channels != dest_format.channels) {
+ buf = pcm_convert_channels_32(channels_buffer,
+ dest_format.channels,
+ src_format.channels,
+ buf, len, &len);
+ if (buf == NULL) {
+ error.Format(pcm_convert_domain,
+ "Conversion from %u to %u channels "
+ "is not implemented",
+ src_format.channels,
+ dest_format.channels);
+ return NULL;
+ }
+ }
+
+ if (src_format.sample_rate != dest_format.sample_rate) {
+ buf = resampler.Resample32(dest_format.channels,
+ src_format.sample_rate, buf, len,
+ dest_format.sample_rate, &len,
+ error);
+ if (buf == NULL)
+ return buf;
+ }
+
+ *dest_size_r = len;
+ return buf;
+}
+
+inline const float *
+PcmConvert::ConvertFloat(const AudioFormat src_format,
+ const void *src_buffer, size_t src_size,
+ const AudioFormat dest_format, size_t *dest_size_r,
+ Error &error)
+{
+ const float *buffer = (const float *)src_buffer;
+ size_t size = src_size;
+
+ assert(dest_format.format == SampleFormat::FLOAT);
+
+ /* convert to float now */
+
+ buffer = pcm_convert_to_float(format_buffer,
+ src_format.format,
+ buffer, size, &size);
+ if (buffer == NULL) {
+ error.Format(pcm_convert_domain,
+ "Conversion from %s to float is not implemented",
+ sample_format_to_string(src_format.format));
+ return NULL;
+ }
+
+ /* convert channels */
+
+ if (src_format.channels != dest_format.channels) {
+ buffer = pcm_convert_channels_float(channels_buffer,
+ dest_format.channels,
+ src_format.channels,
+ buffer, size, &size);
+ if (buffer == NULL) {
+ error.Format(pcm_convert_domain,
+ "Conversion from %u to %u channels "
+ "is not implemented",
+ src_format.channels,
+ dest_format.channels);
+ return NULL;
+ }
+ }
+
+ /* resample with float, because this is the best format for
+ libsamplerate */
+
+ if (src_format.sample_rate != dest_format.sample_rate) {
+ buffer = resampler.ResampleFloat(dest_format.channels,
+ src_format.sample_rate,
+ buffer, size,
+ dest_format.sample_rate,
+ &size, error);
+ if (buffer == NULL)
+ return NULL;
+ }
+
+ *dest_size_r = size;
+ return buffer;
+}
+
+const void *
+PcmConvert::Convert(AudioFormat src_format,
+ const void *src, size_t src_size,
+ const AudioFormat dest_format,
+ size_t *dest_size_r,
+ Error &error)
+{
+ AudioFormat float_format;
+ if (src_format.format == SampleFormat::DSD) {
+ size_t f_size;
+ const float *f = dsd.ToFloat(src_format.channels,
+ false, (const uint8_t *)src,
+ src_size, &f_size);
+ if (f == NULL) {
+ error.Set(pcm_convert_domain,
+ "DSD to PCM conversion failed");
+ return NULL;
+ }
+
+ float_format = src_format;
+ float_format.format = SampleFormat::FLOAT;
+
+ src_format = float_format;
+ src = f;
+ src_size = f_size;
+ }
+
+ switch (dest_format.format) {
+ case SampleFormat::S16:
+ return Convert16(src_format, src, src_size,
+ dest_format, dest_size_r,
+ error);
+
+ case SampleFormat::S24_P32:
+ return Convert24(src_format, src, src_size,
+ dest_format, dest_size_r,
+ error);
+
+ case SampleFormat::S32:
+ return Convert32(src_format, src, src_size,
+ dest_format, dest_size_r,
+ error);
+
+ case SampleFormat::FLOAT:
+ return ConvertFloat(src_format, src, src_size,
+ dest_format, dest_size_r,
+ error);
+
+ default:
+ error.Format(pcm_convert_domain,
+ "PCM conversion to %s is not implemented",
+ sample_format_to_string(dest_format.format));
+ return NULL;
+ }
+}
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