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-rw-r--r--lib/rbcodec/dsp/dsp_sample_input.c334
1 files changed, 334 insertions, 0 deletions
diff --git a/lib/rbcodec/dsp/dsp_sample_input.c b/lib/rbcodec/dsp/dsp_sample_input.c
new file mode 100644
index 0000000..84127e1
--- /dev/null
+++ b/lib/rbcodec/dsp/dsp_sample_input.c
@@ -0,0 +1,334 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Miika Pekkarinen
+ * Copyright (C) 2012 Michael Sevakis
+ *
+ * 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 software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+#include "config.h"
+#include "system.h"
+#include "dsp.h"
+#include "dsp_sample_io.h"
+
+#if 1
+#include <debug.h>
+#else
+#undef DEBUGF
+#define DEBUGF(...)
+#endif
+
+/* The internal format is 32-bit samples, non-interleaved, stereo. This
+ * format is similar to the raw output from several codecs, so no copying is
+ * needed for that case.
+ *
+ * Note that for mono, dst[0] equals dst[1], as there is no point in
+ * processing the same data twice nor should it be done when modifying
+ * samples in-place.
+ *
+ * When conversion is required:
+ * Updates source buffer to point past the samples "consumed" also consuming
+ * that portion of the input buffer and the destination is set to the buffer
+ * of samples for later stages to consume.
+ *
+ * Input operates similarly to how an out-of-place processing stage should
+ * behave.
+ */
+
+extern void dsp_sample_output_init(struct sample_io_data *this);
+extern void dsp_sample_output_flush(struct sample_io_data *this);
+
+/* convert count 16-bit mono to 32-bit mono */
+static void sample_input_mono16(struct sample_io_data *this,
+ struct dsp_buffer **buf_p)
+{
+ struct dsp_buffer *src = *buf_p;
+ struct dsp_buffer *dst = &this->sample_buf;
+
+ *buf_p = dst;
+
+ if (dst->remcount > 0)
+ return; /* data still remains */
+
+ int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
+
+ dst->remcount = count;
+ dst->p32[0] = this->sample_buf_arr[0];
+ dst->p32[1] = this->sample_buf_arr[0];
+ dst->proc_mask = src->proc_mask;
+
+ if (count <= 0)
+ return; /* purged sample_buf */
+
+ const int16_t *s = src->pin[0];
+ int32_t *d = dst->p32[0];
+ const int scale = WORD_SHIFT;
+
+ dsp_advance_buffer_input(src, count, sizeof (int16_t));
+
+ do
+ {
+ *d++ = *s++ << scale;
+ }
+ while (--count > 0);
+}
+
+/* convert count 16-bit interleaved stereo to 32-bit noninterleaved */
+static void sample_input_i_stereo16(struct sample_io_data *this,
+ struct dsp_buffer **buf_p)
+{
+ struct dsp_buffer *src = *buf_p;
+ struct dsp_buffer *dst = &this->sample_buf;
+
+ *buf_p = dst;
+
+ if (dst->remcount > 0)
+ return; /* data still remains */
+
+ int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
+
+ dst->remcount = count;
+ dst->p32[0] = this->sample_buf_arr[0];
+ dst->p32[1] = this->sample_buf_arr[1];
+ dst->proc_mask = src->proc_mask;
+
+ if (count <= 0)
+ return; /* purged sample_buf */
+
+ const int16_t *s = src->pin[0];
+ int32_t *dl = dst->p32[0];
+ int32_t *dr = dst->p32[1];
+ const int scale = WORD_SHIFT;
+
+ dsp_advance_buffer_input(src, count, 2*sizeof (int16_t));
+
+ do
+ {
+ *dl++ = *s++ << scale;
+ *dr++ = *s++ << scale;
+ }
+ while (--count > 0);
+}
+
+/* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */
+static void sample_input_ni_stereo16(struct sample_io_data *this,
+ struct dsp_buffer **buf_p)
+{
+ struct dsp_buffer *src = *buf_p;
+ struct dsp_buffer *dst = &this->sample_buf;
+
+ *buf_p = dst;
+
+ if (dst->remcount > 0)
+ return; /* data still remains */
+
+ int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
+
+ dst->remcount = count;
+ dst->p32[0] = this->sample_buf_arr[0];
+ dst->p32[1] = this->sample_buf_arr[1];
+ dst->proc_mask = src->proc_mask;
+
+ if (count <= 0)
+ return; /* purged sample_buf */
+
+ const int16_t *sl = src->pin[0];
+ const int16_t *sr = src->pin[1];
+ int32_t *dl = dst->p32[0];
+ int32_t *dr = dst->p32[1];
+ const int scale = WORD_SHIFT;
+
+ dsp_advance_buffer_input(src, count, sizeof (int16_t));
+
+ do
+ {
+ *dl++ = *sl++ << scale;
+ *dr++ = *sr++ << scale;
+ }
+ while (--count > 0);
+}
+
+/* convert count 32-bit mono to 32-bit mono */
+static void sample_input_mono32(struct sample_io_data *this,
+ struct dsp_buffer **buf_p)
+{
+ struct dsp_buffer *dst = &this->sample_buf;
+
+ if (dst->remcount > 0)
+ {
+ *buf_p = dst;
+ return; /* data still remains */
+ }
+ /* else no buffer switch */
+
+ struct dsp_buffer *src = *buf_p;
+ src->p32[1] = src->p32[0];
+}
+
+
+/* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */
+static void sample_input_i_stereo32(struct sample_io_data *this,
+ struct dsp_buffer **buf_p)
+{
+ struct dsp_buffer *src = *buf_p;
+ struct dsp_buffer *dst = &this->sample_buf;
+
+ *buf_p = dst;
+
+ if (dst->remcount > 0)
+ return; /* data still remains */
+
+ int count = MIN(src->remcount, SAMPLE_BUF_COUNT);
+
+ dst->remcount = count;
+ dst->p32[0] = this->sample_buf_arr[0];
+ dst->p32[1] = this->sample_buf_arr[1];
+ dst->proc_mask = src->proc_mask;
+
+ if (count <= 0)
+ return; /* purged sample_buf */
+
+ const int32_t *s = src->pin[0];
+ int32_t *dl = dst->p32[0];
+ int32_t *dr = dst->p32[1];
+
+ dsp_advance_buffer_input(src, count, 2*sizeof (int32_t));
+
+ do
+ {
+ *dl++ = *s++;
+ *dr++ = *s++;
+ }
+ while (--count > 0);
+}
+
+/* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */
+static void sample_input_ni_stereo32(struct sample_io_data *this,
+ struct dsp_buffer **buf_p)
+{
+ struct dsp_buffer *dst = &this->sample_buf;
+
+ if (dst->remcount > 0)
+ *buf_p = dst; /* data still remains */
+ /* else no buffer switch */
+}
+
+/* set the to-native sample conversion function based on dsp sample
+ * parameters */
+static void dsp_sample_input_format_change(struct sample_io_data *this,
+ struct dsp_buffer **buf_p)
+{
+ static const sample_input_fn_type fns[STEREO_NUM_MODES][2] =
+ {
+ [STEREO_INTERLEAVED] =
+ { sample_input_i_stereo16,
+ sample_input_i_stereo32 },
+ [STEREO_NONINTERLEAVED] =
+ { sample_input_ni_stereo16,
+ sample_input_ni_stereo32 },
+ [STEREO_MONO] =
+ { sample_input_mono16,
+ sample_input_mono32 },
+ };
+
+ struct dsp_buffer *src = *buf_p;
+ struct dsp_buffer *dst = &this->sample_buf;
+
+ /* Ack configured format change */
+ format_change_ack(&this->format);
+
+ if (dst->remcount > 0)
+ {
+ *buf_p = dst;
+ return; /* data still remains */
+ }
+
+ DSP_PRINT_FORMAT(DSP Input, -1, src->format);
+
+ /* new format - remember it and pass it along */
+ dst->format = src->format;
+ this->input_samples[0] = fns[this->stereo_mode]
+ [this->sample_depth > NATIVE_DEPTH ? 1 : 0];
+
+ this->input_samples[0](this, buf_p);
+
+ if (*buf_p == dst) /* buffer switch? */
+ format_change_ack(&src->format);
+}
+
+static void dsp_sample_input_init(struct sample_io_data *this)
+{
+ this->input_samples[0] = sample_input_ni_stereo32;
+ this->input_samples[1] = dsp_sample_input_format_change;
+}
+
+/* discard the sample buffer */
+static void dsp_sample_input_flush(struct sample_io_data *this)
+{
+ this->sample_buf.remcount = 0;
+}
+
+void dsp_sample_io_configure(struct sample_io_data *this,
+ unsigned int setting,
+ intptr_t value)
+{
+ switch (setting)
+ {
+ case DSP_INIT:
+ dsp_sample_input_init(this);
+ dsp_sample_output_init(this);
+ break;
+
+ case DSP_RESET:
+ /* Reset all sample descriptions to default */
+ format_change_set(&this->format);
+ this->format.num_channels = 2;
+ this->format.frac_bits = WORD_FRACBITS;
+ this->format.output_scale = WORD_FRACBITS + 1 - NATIVE_DEPTH;
+ this->format.frequency = NATIVE_FREQUENCY;
+ this->format.codec_frequency = NATIVE_FREQUENCY;
+ this->sample_depth = NATIVE_DEPTH;
+ this->stereo_mode = STEREO_NONINTERLEAVED;
+ break;
+
+ case DSP_SET_FREQUENCY:
+ value = value > 0 ? value : NATIVE_FREQUENCY;
+ format_change_set(&this->format);
+ this->format.frequency = value;
+ this->format.codec_frequency = value;
+ break;
+
+ case DSP_SET_SAMPLE_DEPTH:
+ format_change_set(&this->format);
+ this->format.frac_bits =
+ value <= NATIVE_DEPTH ? WORD_FRACBITS : value;
+ this->format.output_scale =
+ this->format.frac_bits + 1 - NATIVE_DEPTH;
+ this->sample_depth = value;
+ break;
+
+ case DSP_SET_STEREO_MODE:
+ format_change_set(&this->format);
+ this->format.num_channels = value == STEREO_MONO ? 1 : 2;
+ this->stereo_mode = value;
+ break;
+
+ case DSP_FLUSH:
+ dsp_sample_input_flush(this);
+ dsp_sample_output_flush(this);
+ break;
+ }
+}