pcm_utils: moved code to pcm_resample.c

Separate the resampling code from the rest of pcm_utils.c.  Create two
sub-libraries: pcm_resample_libsamplerate.c and
pcm_resample_fallback.c.

8 files changed, 308 insertions, 176 deletions

diff --git a/configure.ac b/configure.ac
index 8fdec792f..76f4badb8 100644
--- a/configure.ac
+++ b/configure.ac
@@ -390,6 +390,8 @@ if test x$enable_lsr = xyes; then
 	                  [enable_lsr=no;AC_MSG_WARN([libsamplerate not found -- disabling])])
 fi
 
+AM_CONDITIONAL(HAVE_LIBSAMPLERATE, test x$enable_lsr = xyes)
+
 if test x$enable_fifo = xyes; then
 	AC_CHECK_FUNC([mkfifo],
 	              [enable_fifo=yes;AC_DEFINE([HAVE_FIFO], 1, [Define to enable support for writing audio to a FIFO])],
diff --git a/src/Makefile.am b/src/Makefile.am
index f2bf55eb1..47a121484 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -66,6 +66,7 @@ mpd_headers = \
 	path.h \
 	mapper.h \
 	pcm_utils.h \
+	pcm_resample.h \
 	pcm_dither.h \
 	permission.h \
 	player_thread.h \
@@ -146,6 +147,7 @@ mpd_SOURCES = \
 	path.c \
 	mapper.c \
 	pcm_utils.c \
+	pcm_resample.c \
 	pcm_dither.c \
 	permission.c \
 	player_thread.c \
@@ -176,6 +178,12 @@ mpd_SOURCES = \
 	stored_playlist.c \
 	timer.c
 
+if HAVE_LIBSAMPLERATE
+mpd_SOURCES += pcm_resample_libsamplerate.c
+else
+mpd_SOURCES += pcm_resample_fallback.c
+endif
+
 if HAVE_ID3TAG
 mpd_SOURCES += tag_id3.c
 endif
diff --git a/src/pcm_resample.c b/src/pcm_resample.c
new file mode 100644
index 000000000..097e13929
--- /dev/null
+++ b/src/pcm_resample.c
@@ -0,0 +1,26 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2008 Max Kellermann <max@duempel.org>
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include "pcm_resample.h"
+
+#include <string.h>
+
+void pcm_resample_init(struct pcm_resample_state *state)
+{
+	memset(state, 0, sizeof(*state));
+}
diff --git a/src/pcm_resample.h b/src/pcm_resample.h
new file mode 100644
index 000000000..d6001550a
--- /dev/null
+++ b/src/pcm_resample.h
@@ -0,0 +1,63 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * Copyright (C) 2008 Max Kellermann <max@duempel.org>
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#ifndef MPD_PCM_RESAMPLE_H
+#define MPD_PCM_RESAMPLE_H
+
+#include "config.h"
+
+#include <stdint.h>
+#include <stddef.h>
+#include <stdbool.h>
+
+#ifdef HAVE_LIBSAMPLERATE
+#include <samplerate.h>
+#endif
+
+struct pcm_resample_state {
+#ifdef HAVE_LIBSAMPLERATE
+	SRC_STATE *state;
+	SRC_DATA data;
+	size_t data_in_size;
+	size_t data_out_size;
+
+	struct {
+		unsigned src_rate;
+		unsigned dest_rate;
+		uint8_t channels;
+	} prev;
+
+	bool error;
+#else
+	/* struct must not be empty */
+	int dummy;
+#endif
+};
+
+void pcm_resample_init(struct pcm_resample_state *state);
+
+size_t
+pcm_resample_16(uint8_t channels,
+		unsigned src_rate,
+		const int16_t *src_buffer, size_t src_size,
+		unsigned dest_rate,
+		int16_t *dest_buffer, size_t dest_size,
+		struct pcm_resample_state *state);
+
+#endif
diff --git a/src/pcm_resample_fallback.c b/src/pcm_resample_fallback.c
new file mode 100644
index 000000000..4ab586160
--- /dev/null
+++ b/src/pcm_resample_fallback.c
@@ -0,0 +1,61 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * Copyright (C) 2008 Max Kellermann <max@duempel.org>
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include "pcm_resample.h"
+#include "gcc.h"
+
+/* resampling code blatantly ripped from ESD */
+size_t
+pcm_resample_16(uint8_t channels,
+		unsigned src_rate,
+		const int16_t *src_buffer, mpd_unused size_t src_size,
+		unsigned dest_rate,
+		int16_t *dest_buffer, size_t dest_size,
+		mpd_unused struct pcm_resample_state *state)
+{
+	unsigned src_pos, dest_pos = 0;
+	unsigned dest_samples = dest_size / 2;
+	int16_t lsample, rsample;
+
+	switch (channels) {
+	case 1:
+		while (dest_pos < dest_samples) {
+			src_pos = dest_pos * src_rate / dest_rate;
+
+			lsample = src_buffer[src_pos++];
+
+			dest_buffer[dest_pos++] = lsample;
+		}
+		break;
+	case 2:
+		while (dest_pos < dest_samples) {
+			src_pos = dest_pos * src_rate / dest_rate;
+			src_pos &= ~1;
+
+			lsample = src_buffer[src_pos++];
+			rsample = src_buffer[src_pos++];
+
+			dest_buffer[dest_pos++] = lsample;
+			dest_buffer[dest_pos++] = rsample;
+		}
+		break;
+	}
+
+	return dest_size;
+}
diff --git a/src/pcm_resample_libsamplerate.c b/src/pcm_resample_libsamplerate.c
new file mode 100644
index 000000000..5321c0e0b
--- /dev/null
+++ b/src/pcm_resample_libsamplerate.c
@@ -0,0 +1,140 @@
+/* the Music Player Daemon (MPD)
+ * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
+ * Copyright (C) 2008 Max Kellermann <max@duempel.org>
+ * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include "pcm_resample.h"
+#include "conf.h"
+#include "log.h"
+#include "utils.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+static int pcm_resample_get_converter(void)
+{
+	const char *conf = getConfigParamValue(CONF_SAMPLERATE_CONVERTER);
+	long convalgo;
+	char *test;
+	const char *test2;
+	size_t len;
+
+	if (!conf) {
+		convalgo = SRC_SINC_FASTEST;
+		goto out;
+	}
+
+	convalgo = strtol(conf, &test, 10);
+	if (*test == '\0' && src_get_name(convalgo))
+		goto out;
+
+	len = strlen(conf);
+	for (convalgo = 0 ; ; convalgo++) {
+		test2 = src_get_name(convalgo);
+		if (!test2) {
+			convalgo = SRC_SINC_FASTEST;
+			break;
+		}
+		if (strncasecmp(test2, conf, len) == 0)
+			goto out;
+	}
+
+	ERROR("unknown samplerate converter \"%s\"\n", conf);
+out:
+	DEBUG("selecting samplerate converter \"%s\"\n",
+	      src_get_name(convalgo));
+
+	return convalgo;
+}
+
+size_t
+pcm_resample_16(uint8_t channels,
+		unsigned src_rate,
+		const int16_t *src_buffer, size_t src_size,
+		unsigned dest_rate,
+		int16_t *dest_buffer, size_t dest_size,
+		struct pcm_resample_state *state)
+{
+	static int convalgo = -1;
+	SRC_DATA *data = &state->data;
+	size_t data_in_size;
+	size_t data_out_size;
+	int error;
+
+	if (convalgo < 0)
+		convalgo = pcm_resample_get_converter();
+
+	/* (re)set the state/ratio if the in or out format changed */
+	if (channels != state->prev.channels ||
+	    src_rate != state->prev.src_rate ||
+	    dest_rate != state->prev.dest_rate) {
+		state->error = false;
+		state->prev.channels = channels;
+		state->prev.src_rate = src_rate;
+		state->prev.dest_rate = dest_rate;
+
+		if (state->state)
+			state->state = src_delete(state->state);
+
+		state->state = src_new(convalgo, channels, &error);
+		if (!state->state) {
+			ERROR("cannot create new libsamplerate state: %s\n",
+			      src_strerror(error));
+			state->error = true;
+			return 0;
+		}
+
+		data->src_ratio = (double)dest_rate / (double)src_rate;
+		DEBUG("setting samplerate conversion ratio to %.2lf\n",
+		      data->src_ratio);
+		src_set_ratio(state->state, data->src_ratio);
+	}
+
+	/* there was an error previously, and nothing has changed */
+	if (state->error)
+		return 0;
+
+	data->input_frames = src_size / 2 / channels;
+	data_in_size = data->input_frames * sizeof(float) * channels;
+	if (data_in_size > state->data_in_size) {
+		state->data_in_size = data_in_size;
+		data->data_in = xrealloc(data->data_in, data_in_size);
+	}
+
+	data->output_frames = dest_size / 2 / channels;
+	data_out_size = data->output_frames * sizeof(float) * channels;
+	if (data_out_size > state->data_out_size) {
+		state->data_out_size = data_out_size;
+		data->data_out = xrealloc(data->data_out, data_out_size);
+	}
+
+	src_short_to_float_array(src_buffer, data->data_in,
+				 data->input_frames * channels);
+
+	error = src_process(state->state, data);
+	if (error) {
+		ERROR("error processing samples with libsamplerate: %s\n",
+		      src_strerror(error));
+		state->error = true;
+		return 0;
+	}
+
+	src_float_to_short_array(data->data_out, dest_buffer,
+				 data->output_frames_gen * channels);
+
+	return data->output_frames_gen * 2 * channels;
+}
diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index 8d5bed990..6d76aba90 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -209,166 +209,10 @@ void pcm_convert_init(struct pcm_convert_state *state)
 {
 	memset(state, 0, sizeof(*state));
 
+	pcm_resample_init(&state->resample);
 	pcm_dither_24_init(&state->dither);
 }
 
-#ifdef HAVE_LIBSAMPLERATE
-static int pcm_resample_get_converter(void)
-{
-	const char *conf = getConfigParamValue(CONF_SAMPLERATE_CONVERTER);
-	long convalgo;
-	char *test;
-	const char *test2;
-	size_t len;
-
-	if (!conf) {
-		convalgo = SRC_SINC_FASTEST;
-		goto out;
-	}
-
-	convalgo = strtol(conf, &test, 10);
-	if (*test == '\0' && src_get_name(convalgo))
-		goto out;
-
-	len = strlen(conf);
-	for (convalgo = 0 ; ; convalgo++) {
-		test2 = src_get_name(convalgo);
-		if (!test2) {
-			convalgo = SRC_SINC_FASTEST;
-			break;
-		}
-		if (strncasecmp(test2, conf, len) == 0)
-			goto out;
-	}
-
-	ERROR("unknown samplerate converter \"%s\"\n", conf);
-out:
-	DEBUG("selecting samplerate converter \"%s\"\n",
-	      src_get_name(convalgo));
-
-	return convalgo;
-}
-#endif
-
-#ifdef HAVE_LIBSAMPLERATE
-static size_t pcm_resample(int8_t channels, uint32_t inSampleRate,
-			   const int16_t *src, size_t src_size,
-			   uint32_t outSampleRate, int16_t *outBuffer,
-			   size_t outSize,
-			   struct pcm_convert_state *convState)
-{
-	static int convalgo = -1;
-	SRC_DATA *data = &convState->data;
-	size_t dataInSize;
-	size_t dataOutSize;
-	int error;
-
-	if (convalgo < 0)
-		convalgo = pcm_resample_get_converter();
-
-	/* (re)set the state/ratio if the in or out format changed */
-	if ((channels != convState->lastChannels) ||
-	    (inSampleRate != convState->lastInSampleRate) ||
-	    (outSampleRate != convState->lastOutSampleRate)) {
-		convState->error = 0;
-		convState->lastChannels = channels;
-		convState->lastInSampleRate = inSampleRate;
-		convState->lastOutSampleRate = outSampleRate;
-
-		if (convState->state)
-			convState->state = src_delete(convState->state);
-
-		convState->state = src_new(convalgo, channels, &error);
-		if (!convState->state) {
-			ERROR("cannot create new libsamplerate state: %s\n",
-			      src_strerror(error));
-			convState->error = 1;
-			return 0;
-		}
-		
-		data->src_ratio = (double)outSampleRate / (double)inSampleRate;
-		DEBUG("setting samplerate conversion ratio to %.2lf\n",
-		      data->src_ratio);
-		src_set_ratio(convState->state, data->src_ratio);
-	}
-
-	/* there was an error previously, and nothing has changed */
-	if (convState->error)
-		return 0;
-
-	data->input_frames = src_size / 2 / channels;
-	dataInSize = data->input_frames * sizeof(float) * channels;
-	if (dataInSize > convState->dataInSize) {
-		convState->dataInSize = dataInSize;
-		data->data_in = xrealloc(data->data_in, dataInSize);
-	}
-
-	data->output_frames = outSize / 2 / channels;
-	dataOutSize = data->output_frames * sizeof(float) * channels;
-	if (dataOutSize > convState->dataOutSize) {
-		convState->dataOutSize = dataOutSize;
-		data->data_out = xrealloc(data->data_out, dataOutSize);
-	}
-
-	src_short_to_float_array((const short *)src, data->data_in,
-	                         data->input_frames * channels);
-
-	error = src_process(convState->state, data);
-	if (error) {
-		ERROR("error processing samples with libsamplerate: %s\n",
-		      src_strerror(error));
-		convState->error = 1;
-		return 0;
-	}
-
-	src_float_to_short_array(data->data_out, (short *)outBuffer,
-	                         data->output_frames_gen * channels);
-
-	return data->output_frames_gen * 2 * channels;
-}
-#else /* !HAVE_LIBSAMPLERATE */
-/* resampling code blatantly ripped from ESD */
-static size_t pcm_resample(int8_t channels, uint32_t inSampleRate,
-			   const int16_t *src, mpd_unused size_t src_size,
-			   uint32_t outSampleRate, char *outBuffer,
-			   size_t outSize,
-			   mpd_unused struct pcm_convert_state *convState)
-{
-	uint32_t rd_dat = 0;
-	uint32_t wr_dat = 0;
-	const int16_t *in = (const int16_t *)src;
-	int16_t *out = (int16_t *)outBuffer;
-	uint32_t nlen = outSize / 2;
-	int16_t lsample, rsample;
-
-	switch (channels) {
-	case 1:
-		while (wr_dat < nlen) {
-			rd_dat = wr_dat * inSampleRate / outSampleRate;
-
-			lsample = in[rd_dat++];
-
-			out[wr_dat++] = lsample;
-		}
-		break;
-	case 2:
-		while (wr_dat < nlen) {
-			rd_dat = wr_dat * inSampleRate / outSampleRate;
-			rd_dat &= ~1;
-
-			lsample = in[rd_dat++];
-			rsample = in[rd_dat++];
-
-			out[wr_dat++] = lsample;
-			out[wr_dat++] = rsample;
-		}
-		break;
-	}
-
-	return outSize;
-}
-#endif /* !HAVE_LIBSAMPLERATE */
-
 static void
 pcm_convert_channels_1_to_2(int16_t *dest, const int16_t *src,
 			    unsigned num_frames)
@@ -542,11 +386,11 @@ size_t pcm_convert(const struct audio_format *inFormat,
 		assert(dest_size >= len);
 		memcpy(outBuffer, buf, len);
 	} else {
-		len = pcm_resample(outFormat->channels,
-				   inFormat->sample_rate, buf, len,
-				   outFormat->sample_rate,
-				   (int16_t*)outBuffer,
-				   dest_size, convState);
+		len = pcm_resample_16(outFormat->channels,
+				      inFormat->sample_rate, buf, len,
+				      outFormat->sample_rate,
+				      (int16_t*)outBuffer,
+				      dest_size, &convState->resample);
 		if (len == 0)
 			exit(EXIT_FAILURE);
 	}
diff --git a/src/pcm_utils.h b/src/pcm_utils.h
index 835ba187f..3599b7076 100644
--- a/src/pcm_utils.h
+++ b/src/pcm_utils.h
@@ -19,28 +19,16 @@
 #ifndef PCM_UTILS_H
 #define PCM_UTILS_H
 
-#include "../config.h"
+#include "pcm_resample.h"
 #include "pcm_dither.h"
 
 #include <stdint.h>
 #include <stddef.h>
 
-#ifdef HAVE_LIBSAMPLERATE
-#include <samplerate.h>
-#endif
-
 struct audio_format;
 
 struct pcm_convert_state {
-#ifdef HAVE_LIBSAMPLERATE
-	SRC_STATE *state;
-	SRC_DATA data;
-	size_t dataInSize;
-	size_t dataOutSize;
-	int8_t lastChannels;
-	uint32_t lastInSampleRate;
-	uint32_t lastOutSampleRate;
-#endif
+	struct pcm_resample_state resample;
 
 	struct pcm_dither_24 dither;