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-rw-r--r--src/pulsecore/resampler/ffmpeg.c132
-rw-r--r--src/pulsecore/resampler/libsamplerate.c100
-rw-r--r--src/pulsecore/resampler/peaks.c161
-rw-r--r--src/pulsecore/resampler/soxr.c168
-rw-r--r--src/pulsecore/resampler/speex.c178
-rw-r--r--src/pulsecore/resampler/trivial.c100
6 files changed, 839 insertions, 0 deletions
diff --git a/src/pulsecore/resampler/ffmpeg.c b/src/pulsecore/resampler/ffmpeg.c
new file mode 100644
index 0000000..388b555
--- /dev/null
+++ b/src/pulsecore/resampler/ffmpeg.c
@@ -0,0 +1,132 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2004-2006 Lennart Poettering
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio 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 Lesser General Public License
+ along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/xmalloc.h>
+#include "pulsecore/ffmpeg/avcodec.h"
+
+#include <pulsecore/resampler.h>
+
+struct ffmpeg_data { /* data specific to ffmpeg */
+ struct AVResampleContext *state;
+};
+
+static unsigned ffmpeg_resample(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
+ unsigned used_frames = 0, c;
+ int previous_consumed_frames = -1;
+ struct ffmpeg_data *ffmpeg_data;
+
+ pa_assert(r);
+ pa_assert(input);
+ pa_assert(output);
+ pa_assert(out_n_frames);
+
+ ffmpeg_data = r->impl.data;
+
+ for (c = 0; c < r->work_channels; c++) {
+ unsigned u;
+ pa_memblock *b, *w;
+ int16_t *p, *t, *k, *q, *s;
+ int consumed_frames;
+
+ /* Allocate a new block */
+ b = pa_memblock_new(r->mempool, in_n_frames * sizeof(int16_t));
+ p = pa_memblock_acquire(b);
+
+ /* Now copy the input data, splitting up channels */
+ t = (int16_t*) pa_memblock_acquire_chunk(input) + c;
+ k = p;
+ for (u = 0; u < in_n_frames; u++) {
+ *k = *t;
+ t += r->work_channels;
+ k ++;
+ }
+ pa_memblock_release(input->memblock);
+
+ /* Allocate buffer for the result */
+ w = pa_memblock_new(r->mempool, *out_n_frames * sizeof(int16_t));
+ q = pa_memblock_acquire(w);
+
+ /* Now, resample */
+ used_frames = (unsigned) av_resample(ffmpeg_data->state,
+ q, p,
+ &consumed_frames,
+ (int) in_n_frames, (int) *out_n_frames,
+ c >= (unsigned) (r->work_channels-1));
+
+ pa_memblock_release(b);
+ pa_memblock_unref(b);
+
+ pa_assert(consumed_frames <= (int) in_n_frames);
+ pa_assert(previous_consumed_frames == -1 || consumed_frames == previous_consumed_frames);
+ previous_consumed_frames = consumed_frames;
+
+ /* And place the results in the output buffer */
+ s = (int16_t *) pa_memblock_acquire_chunk(output) + c;
+ for (u = 0; u < used_frames; u++) {
+ *s = *q;
+ q++;
+ s += r->work_channels;
+ }
+ pa_memblock_release(output->memblock);
+ pa_memblock_release(w);
+ pa_memblock_unref(w);
+ }
+
+ *out_n_frames = used_frames;
+
+ return in_n_frames - previous_consumed_frames;
+}
+
+static void ffmpeg_free(pa_resampler *r) {
+ struct ffmpeg_data *ffmpeg_data;
+
+ pa_assert(r);
+
+ ffmpeg_data = r->impl.data;
+ if (ffmpeg_data->state)
+ av_resample_close(ffmpeg_data->state);
+}
+
+int pa_resampler_ffmpeg_init(pa_resampler *r) {
+ struct ffmpeg_data *ffmpeg_data;
+
+ pa_assert(r);
+
+ ffmpeg_data = pa_xnew(struct ffmpeg_data, 1);
+
+ /* We could probably implement different quality levels by
+ * adjusting the filter parameters here. However, ffmpeg
+ * internally only uses these hardcoded values, so let's use them
+ * here for now as well until ffmpeg makes this configurable. */
+
+ if (!(ffmpeg_data->state = av_resample_init((int) r->o_ss.rate, (int) r->i_ss.rate, 16, 10, 0, 0.8))) {
+ pa_xfree(ffmpeg_data);
+ return -1;
+ }
+
+ r->impl.free = ffmpeg_free;
+ r->impl.resample = ffmpeg_resample;
+ r->impl.data = (void *) ffmpeg_data;
+
+ return 0;
+}
diff --git a/src/pulsecore/resampler/libsamplerate.c b/src/pulsecore/resampler/libsamplerate.c
new file mode 100644
index 0000000..06704fe
--- /dev/null
+++ b/src/pulsecore/resampler/libsamplerate.c
@@ -0,0 +1,100 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2004-2006 Lennart Poettering
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio 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 Lesser General Public License
+ along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <samplerate.h>
+
+#include <pulsecore/resampler.h>
+
+static unsigned libsamplerate_resample(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
+ SRC_DATA data;
+ SRC_STATE *state;
+
+ pa_assert(r);
+ pa_assert(input);
+ pa_assert(output);
+ pa_assert(out_n_frames);
+
+ state = r->impl.data;
+ memset(&data, 0, sizeof(data));
+
+ data.data_in = pa_memblock_acquire_chunk(input);
+ data.input_frames = (long int) in_n_frames;
+
+ data.data_out = pa_memblock_acquire_chunk(output);
+ data.output_frames = (long int) *out_n_frames;
+
+ data.src_ratio = (double) r->o_ss.rate / r->i_ss.rate;
+ data.end_of_input = 0;
+
+ pa_assert_se(src_process(state, &data) == 0);
+
+ pa_memblock_release(input->memblock);
+ pa_memblock_release(output->memblock);
+
+ *out_n_frames = (unsigned) data.output_frames_gen;
+
+ return in_n_frames - data.input_frames_used;
+}
+
+static void libsamplerate_update_rates(pa_resampler *r) {
+ SRC_STATE *state;
+ pa_assert(r);
+
+ state = r->impl.data;
+ pa_assert_se(src_set_ratio(state, (double) r->o_ss.rate / r->i_ss.rate) == 0);
+}
+
+static void libsamplerate_reset(pa_resampler *r) {
+ SRC_STATE *state;
+ pa_assert(r);
+
+ state = r->impl.data;
+ pa_assert_se(src_reset(state) == 0);
+}
+
+static void libsamplerate_free(pa_resampler *r) {
+ SRC_STATE *state;
+ pa_assert(r);
+
+ state = r->impl.data;
+ if (state)
+ src_delete(state);
+}
+
+int pa_resampler_libsamplerate_init(pa_resampler *r) {
+ int err;
+ SRC_STATE *state;
+
+ pa_assert(r);
+
+ if (!(state = src_new(r->method, r->work_channels, &err)))
+ return -1;
+
+ r->impl.free = libsamplerate_free;
+ r->impl.update_rates = libsamplerate_update_rates;
+ r->impl.resample = libsamplerate_resample;
+ r->impl.reset = libsamplerate_reset;
+ r->impl.data = state;
+
+ return 0;
+}
diff --git a/src/pulsecore/resampler/peaks.c b/src/pulsecore/resampler/peaks.c
new file mode 100644
index 0000000..c9b808e
--- /dev/null
+++ b/src/pulsecore/resampler/peaks.c
@@ -0,0 +1,161 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2004-2006 Lennart Poettering
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio 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 Lesser General Public License
+ along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/xmalloc.h>
+#include <math.h>
+
+#include <pulsecore/resampler.h>
+
+struct peaks_data { /* data specific to the peak finder pseudo resampler */
+ unsigned o_counter;
+ unsigned i_counter;
+
+ float max_f[PA_CHANNELS_MAX];
+ int16_t max_i[PA_CHANNELS_MAX];
+};
+
+static unsigned peaks_resample(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
+ unsigned c, o_index = 0;
+ unsigned i, i_end = 0;
+ void *src, *dst;
+ struct peaks_data *peaks_data;
+
+ pa_assert(r);
+ pa_assert(input);
+ pa_assert(output);
+ pa_assert(out_n_frames);
+
+ peaks_data = r->impl.data;
+ src = pa_memblock_acquire_chunk(input);
+ dst = pa_memblock_acquire_chunk(output);
+
+ i = ((uint64_t) peaks_data->o_counter * r->i_ss.rate) / r->o_ss.rate;
+ i = i > peaks_data->i_counter ? i - peaks_data->i_counter : 0;
+
+ while (i_end < in_n_frames) {
+ i_end = ((uint64_t) (peaks_data->o_counter + 1) * r->i_ss.rate) / r->o_ss.rate;
+ i_end = i_end > peaks_data->i_counter ? i_end - peaks_data->i_counter : 0;
+
+ pa_assert_fp(o_index * r->w_fz < pa_memblock_get_length(output->memblock));
+
+ /* 1ch float is treated separately, because that is the common case */
+ if (r->work_channels == 1 && r->work_format == PA_SAMPLE_FLOAT32NE) {
+ float *s = (float*) src + i;
+ float *d = (float*) dst + o_index;
+
+ for (; i < i_end && i < in_n_frames; i++) {
+ float n = fabsf(*s++);
+
+ if (n > peaks_data->max_f[0])
+ peaks_data->max_f[0] = n;
+ }
+
+ if (i == i_end) {
+ *d = peaks_data->max_f[0];
+ peaks_data->max_f[0] = 0;
+ o_index++, peaks_data->o_counter++;
+ }
+ } else if (r->work_format == PA_SAMPLE_S16NE) {
+ int16_t *s = (int16_t*) src + r->work_channels * i;
+ int16_t *d = (int16_t*) dst + r->work_channels * o_index;
+
+ for (; i < i_end && i < in_n_frames; i++)
+ for (c = 0; c < r->work_channels; c++) {
+ int16_t n = abs(*s++);
+
+ if (n > peaks_data->max_i[c])
+ peaks_data->max_i[c] = n;
+ }
+
+ if (i == i_end) {
+ for (c = 0; c < r->work_channels; c++, d++) {
+ *d = peaks_data->max_i[c];
+ peaks_data->max_i[c] = 0;
+ }
+ o_index++, peaks_data->o_counter++;
+ }
+ } else {
+ float *s = (float*) src + r->work_channels * i;
+ float *d = (float*) dst + r->work_channels * o_index;
+
+ for (; i < i_end && i < in_n_frames; i++)
+ for (c = 0; c < r->work_channels; c++) {
+ float n = fabsf(*s++);
+
+ if (n > peaks_data->max_f[c])
+ peaks_data->max_f[c] = n;
+ }
+
+ if (i == i_end) {
+ for (c = 0; c < r->work_channels; c++, d++) {
+ *d = peaks_data->max_f[c];
+ peaks_data->max_f[c] = 0;
+ }
+ o_index++, peaks_data->o_counter++;
+ }
+ }
+ }
+
+ pa_memblock_release(input->memblock);
+ pa_memblock_release(output->memblock);
+
+ *out_n_frames = o_index;
+
+ peaks_data->i_counter += in_n_frames;
+
+ /* Normalize counters */
+ while (peaks_data->i_counter >= r->i_ss.rate) {
+ pa_assert(peaks_data->o_counter >= r->o_ss.rate);
+
+ peaks_data->i_counter -= r->i_ss.rate;
+ peaks_data->o_counter -= r->o_ss.rate;
+ }
+
+ return 0;
+}
+
+static void peaks_update_rates_or_reset(pa_resampler *r) {
+ struct peaks_data *peaks_data;
+ pa_assert(r);
+
+ peaks_data = r->impl.data;
+
+ peaks_data->i_counter = 0;
+ peaks_data->o_counter = 0;
+}
+
+int pa_resampler_peaks_init(pa_resampler*r) {
+ struct peaks_data *peaks_data;
+ pa_assert(r);
+ pa_assert(r->i_ss.rate >= r->o_ss.rate);
+ pa_assert(r->work_format == PA_SAMPLE_S16NE || r->work_format == PA_SAMPLE_FLOAT32NE);
+
+ peaks_data = pa_xnew0(struct peaks_data, 1);
+
+ r->impl.resample = peaks_resample;
+ r->impl.update_rates = peaks_update_rates_or_reset;
+ r->impl.reset = peaks_update_rates_or_reset;
+ r->impl.data = peaks_data;
+
+ return 0;
+}
diff --git a/src/pulsecore/resampler/soxr.c b/src/pulsecore/resampler/soxr.c
new file mode 100644
index 0000000..b1b2e19
--- /dev/null
+++ b/src/pulsecore/resampler/soxr.c
@@ -0,0 +1,168 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2014, 2015 Andrey Semashev
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio 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 Lesser General Public License
+ along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stddef.h>
+#include <soxr.h>
+
+#include <pulsecore/resampler.h>
+
+static unsigned resampler_soxr_resample(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames,
+ pa_memchunk *output, unsigned *out_n_frames) {
+ soxr_t state;
+ void *in, *out;
+ size_t consumed = 0, produced = 0;
+
+ pa_assert(r);
+ pa_assert(input);
+ pa_assert(output);
+ pa_assert(out_n_frames);
+
+ state = r->impl.data;
+ pa_assert(state);
+
+ in = pa_memblock_acquire_chunk(input);
+ out = pa_memblock_acquire_chunk(output);
+
+ pa_assert_se(soxr_process(state, in, in_n_frames, &consumed, out, *out_n_frames, &produced) == 0);
+
+ pa_memblock_release(input->memblock);
+ pa_memblock_release(output->memblock);
+
+ *out_n_frames = produced;
+
+ return in_n_frames - consumed;
+}
+
+static void resampler_soxr_free(pa_resampler *r) {
+ pa_assert(r);
+
+ if (!r->impl.data)
+ return;
+
+ soxr_delete(r->impl.data);
+ r->impl.data = NULL;
+}
+
+static void resampler_soxr_reset(pa_resampler *r) {
+#if SOXR_THIS_VERSION >= SOXR_VERSION(0, 1, 2)
+ pa_assert(r);
+
+ soxr_clear(r->impl.data);
+#else
+ /* With libsoxr prior to 0.1.2 soxr_clear() makes soxr_process() crash afterwards,
+ * so don't use this function and re-create the context instead. */
+ soxr_t old_state;
+
+ pa_assert(r);
+
+ old_state = r->impl.data;
+ r->impl.data = NULL;
+
+ if (pa_resampler_soxr_init(r) == 0) {
+ if (old_state)
+ soxr_delete(old_state);
+ } else {
+ r->impl.data = old_state;
+ pa_log_error("Failed to reset libsoxr context");
+ }
+#endif
+}
+
+static void resampler_soxr_update_rates(pa_resampler *r) {
+ soxr_t old_state;
+
+ pa_assert(r);
+
+ /* There is no update method in libsoxr,
+ * so just re-create the resampler context */
+
+ old_state = r->impl.data;
+ r->impl.data = NULL;
+
+ if (pa_resampler_soxr_init(r) == 0) {
+ if (old_state)
+ soxr_delete(old_state);
+ } else {
+ r->impl.data = old_state;
+ pa_log_error("Failed to update libsoxr sample rates");
+ }
+}
+
+int pa_resampler_soxr_init(pa_resampler *r) {
+ soxr_t state;
+ soxr_datatype_t io_format;
+ soxr_io_spec_t io_spec;
+ soxr_runtime_spec_t runtime_spec;
+ unsigned long quality_recipe;
+ soxr_quality_spec_t quality;
+ soxr_error_t err = NULL;
+
+ pa_assert(r);
+
+ switch (r->work_format) {
+ case PA_SAMPLE_S16NE:
+ io_format = SOXR_INT16_I;
+ break;
+ case PA_SAMPLE_FLOAT32NE:
+ io_format = SOXR_FLOAT32_I;
+ break;
+ default:
+ pa_assert_not_reached();
+ }
+
+ io_spec = soxr_io_spec(io_format, io_format);
+
+ /* Resample in one thread. Multithreading makes
+ * performance worse with small chunks of audio. */
+ runtime_spec = soxr_runtime_spec(1);
+
+ switch (r->method) {
+ case PA_RESAMPLER_SOXR_MQ:
+ quality_recipe = SOXR_MQ | SOXR_LINEAR_PHASE;
+ break;
+ case PA_RESAMPLER_SOXR_HQ:
+ quality_recipe = SOXR_HQ | SOXR_LINEAR_PHASE;
+ break;
+ case PA_RESAMPLER_SOXR_VHQ:
+ quality_recipe = SOXR_VHQ | SOXR_LINEAR_PHASE;
+ break;
+ default:
+ pa_assert_not_reached();
+ }
+
+ quality = soxr_quality_spec(quality_recipe, 0);
+
+ state = soxr_create(r->i_ss.rate, r->o_ss.rate, r->work_channels, &err, &io_spec, &quality, &runtime_spec);
+ if (!state) {
+ pa_log_error("Failed to create libsoxr resampler context: %s.", (err ? err : "[unknown error]"));
+ return -1;
+ }
+
+ r->impl.free = resampler_soxr_free;
+ r->impl.reset = resampler_soxr_reset;
+ r->impl.update_rates = resampler_soxr_update_rates;
+ r->impl.resample = resampler_soxr_resample;
+ r->impl.data = state;
+
+ return 0;
+}
diff --git a/src/pulsecore/resampler/speex.c b/src/pulsecore/resampler/speex.c
new file mode 100644
index 0000000..66387e5
--- /dev/null
+++ b/src/pulsecore/resampler/speex.c
@@ -0,0 +1,178 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2004-2006 Lennart Poettering
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio 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 Lesser General Public License
+ along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <speex/speex_resampler.h>
+#include <math.h>
+
+#include <pulsecore/once.h>
+#include <pulsecore/resampler.h>
+
+bool pa_speex_is_fixed_point(void) {
+ static bool result = false;
+ PA_ONCE_BEGIN {
+ float f_out = -1.0f, f_in = 1.0f;
+ spx_uint32_t in_len = 1, out_len = 1;
+ SpeexResamplerState *s;
+
+ pa_assert_se(s = speex_resampler_init(1, 1, 1,
+ SPEEX_RESAMPLER_QUALITY_MIN, NULL));
+
+ /* feed one sample that is too soft for fixed-point speex */
+ pa_assert_se(speex_resampler_process_float(s, 0, &f_in, &in_len,
+ &f_out, &out_len) == RESAMPLER_ERR_SUCCESS);
+
+ /* expecting sample has been processed, one sample output */
+ pa_assert_se(in_len == 1 && out_len == 1);
+
+ /* speex compiled with --enable-fixed-point will output 0.0 due to insufficient precision */
+ if (fabsf(f_out) < 0.00001f)
+ result = true;
+
+ speex_resampler_destroy(s);
+ } PA_ONCE_END;
+ return result;
+}
+
+
+static unsigned speex_resample_float(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
+ float *in, *out;
+ uint32_t inf = in_n_frames, outf = *out_n_frames;
+ SpeexResamplerState *state;
+
+ pa_assert(r);
+ pa_assert(input);
+ pa_assert(output);
+ pa_assert(out_n_frames);
+
+ state = r->impl.data;
+
+ in = pa_memblock_acquire_chunk(input);
+ out = pa_memblock_acquire_chunk(output);
+
+ /* Strictly speaking, speex resampler expects its input
+ * to be normalized to the [-32768.0 .. 32767.0] range.
+ * This matters if speex has been compiled with --enable-fixed-point,
+ * because such speex will round the samples to the nearest
+ * integer. speex with --enable-fixed-point is therefore incompatible
+ * with PulseAudio's floating-point sample range [-1 .. 1]. speex
+ * without --enable-fixed-point works fine with this range.
+ * Care has been taken to call speex_resample_float() only
+ * for speex compiled without --enable-fixed-point.
+ */
+ pa_assert_se(speex_resampler_process_interleaved_float(state, in, &inf, out, &outf) == 0);
+
+ pa_memblock_release(input->memblock);
+ pa_memblock_release(output->memblock);
+
+ pa_assert(inf == in_n_frames);
+ *out_n_frames = outf;
+
+ return 0;
+}
+
+static unsigned speex_resample_int(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
+ int16_t *in, *out;
+ uint32_t inf = in_n_frames, outf = *out_n_frames;
+ SpeexResamplerState *state;
+
+ pa_assert(r);
+ pa_assert(input);
+ pa_assert(output);
+ pa_assert(out_n_frames);
+
+ state = r->impl.data;
+
+ in = pa_memblock_acquire_chunk(input);
+ out = pa_memblock_acquire_chunk(output);
+
+ pa_assert_se(speex_resampler_process_interleaved_int(state, in, &inf, out, &outf) == 0);
+
+ pa_memblock_release(input->memblock);
+ pa_memblock_release(output->memblock);
+
+ pa_assert(inf == in_n_frames);
+ *out_n_frames = outf;
+
+ return 0;
+}
+
+static void speex_update_rates(pa_resampler *r) {
+ SpeexResamplerState *state;
+ pa_assert(r);
+
+ state = r->impl.data;
+
+ pa_assert_se(speex_resampler_set_rate(state, r->i_ss.rate, r->o_ss.rate) == 0);
+}
+
+static void speex_reset(pa_resampler *r) {
+ SpeexResamplerState *state;
+ pa_assert(r);
+
+ state = r->impl.data;
+
+ pa_assert_se(speex_resampler_reset_mem(state) == 0);
+}
+
+static void speex_free(pa_resampler *r) {
+ SpeexResamplerState *state;
+ pa_assert(r);
+
+ state = r->impl.data;
+ if (!state)
+ return;
+
+ speex_resampler_destroy(state);
+}
+
+int pa_resampler_speex_init(pa_resampler *r) {
+ int q, err;
+ SpeexResamplerState *state;
+
+ pa_assert(r);
+
+ r->impl.free = speex_free;
+ r->impl.update_rates = speex_update_rates;
+ r->impl.reset = speex_reset;
+
+ if (r->method >= PA_RESAMPLER_SPEEX_FIXED_BASE && r->method <= PA_RESAMPLER_SPEEX_FIXED_MAX) {
+
+ q = r->method - PA_RESAMPLER_SPEEX_FIXED_BASE;
+ r->impl.resample = speex_resample_int;
+
+ } else {
+ pa_assert(r->method >= PA_RESAMPLER_SPEEX_FLOAT_BASE && r->method <= PA_RESAMPLER_SPEEX_FLOAT_MAX);
+
+ q = r->method - PA_RESAMPLER_SPEEX_FLOAT_BASE;
+ r->impl.resample = speex_resample_float;
+ }
+
+ pa_log_info("Choosing speex quality setting %i.", q);
+
+ if (!(state = speex_resampler_init(r->work_channels, r->i_ss.rate, r->o_ss.rate, q, &err)))
+ return -1;
+
+ r->impl.data = state;
+
+ return 0;
+}
diff --git a/src/pulsecore/resampler/trivial.c b/src/pulsecore/resampler/trivial.c
new file mode 100644
index 0000000..14e7ef3
--- /dev/null
+++ b/src/pulsecore/resampler/trivial.c
@@ -0,0 +1,100 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2004-2006 Lennart Poettering
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio 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 Lesser General Public License
+ along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/xmalloc.h>
+
+#include <pulsecore/resampler.h>
+
+struct trivial_data { /* data specific to the trivial resampler */
+ unsigned o_counter;
+ unsigned i_counter;
+};
+
+static unsigned trivial_resample(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
+ unsigned i_index, o_index;
+ void *src, *dst;
+ struct trivial_data *trivial_data;
+
+ pa_assert(r);
+ pa_assert(input);
+ pa_assert(output);
+ pa_assert(out_n_frames);
+
+ trivial_data = r->impl.data;
+
+ src = pa_memblock_acquire_chunk(input);
+ dst = pa_memblock_acquire_chunk(output);
+
+ for (o_index = 0;; o_index++, trivial_data->o_counter++) {
+ i_index = ((uint64_t) trivial_data->o_counter * r->i_ss.rate) / r->o_ss.rate;
+ i_index = i_index > trivial_data->i_counter ? i_index - trivial_data->i_counter : 0;
+
+ if (i_index >= in_n_frames)
+ break;
+
+ pa_assert_fp(o_index * r->w_fz < pa_memblock_get_length(output->memblock));
+
+ memcpy((uint8_t*) dst + r->w_fz * o_index, (uint8_t*) src + r->w_fz * i_index, (int) r->w_fz);
+ }
+
+ pa_memblock_release(input->memblock);
+ pa_memblock_release(output->memblock);
+
+ *out_n_frames = o_index;
+
+ trivial_data->i_counter += in_n_frames;
+
+ /* Normalize counters */
+ while (trivial_data->i_counter >= r->i_ss.rate) {
+ pa_assert(trivial_data->o_counter >= r->o_ss.rate);
+
+ trivial_data->i_counter -= r->i_ss.rate;
+ trivial_data->o_counter -= r->o_ss.rate;
+ }
+
+ return 0;
+}
+
+static void trivial_update_rates_or_reset(pa_resampler *r) {
+ struct trivial_data *trivial_data;
+ pa_assert(r);
+
+ trivial_data = r->impl.data;
+
+ trivial_data->i_counter = 0;
+ trivial_data->o_counter = 0;
+}
+
+int pa_resampler_trivial_init(pa_resampler *r) {
+ struct trivial_data *trivial_data;
+ pa_assert(r);
+
+ trivial_data = pa_xnew0(struct trivial_data, 1);
+
+ r->impl.resample = trivial_resample;
+ r->impl.update_rates = trivial_update_rates_or_reset;
+ r->impl.reset = trivial_update_rates_or_reset;
+ r->impl.data = trivial_data;
+
+ return 0;
+}
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-22 03:46:14 +0000