[pulseaudio-discuss] [PATCH 1/3] resampler: Split the resampler implementations into separate files

Peter Meerwald pmeerw at pmeerw.net
Mon Aug 4 05:40:12 PDT 2014


Signed-off-by: Peter Meerwald <pmeerw at pmeerw.net>
Signed-off-by: poljar (Damir Jelić) <poljarinho at gmail.com>

--

changes over https://github.com/poljar/pulseaudio/commits/resampler_implementations_v2:
rebased
---
 src/Makefile.am                         |  18 +-
 src/pulsecore/resampler.c               | 634 ++------------------------------
 src/pulsecore/resampler.h               |  50 +++
 src/pulsecore/resampler/ffmpeg.c        | 132 +++++++
 src/pulsecore/resampler/libsamplerate.c | 102 +++++
 src/pulsecore/resampler/peaks.c         | 163 ++++++++
 src/pulsecore/resampler/speex.c         | 151 ++++++++
 src/pulsecore/resampler/trivial.c       | 102 +++++
 8 files changed, 748 insertions(+), 604 deletions(-)
 create mode 100644 src/pulsecore/resampler/ffmpeg.c
 create mode 100644 src/pulsecore/resampler/libsamplerate.c
 create mode 100644 src/pulsecore/resampler/peaks.c
 create mode 100644 src/pulsecore/resampler/speex.c
 create mode 100644 src/pulsecore/resampler/trivial.c

diff --git a/src/Makefile.am b/src/Makefile.am
index 5924bd8..21eb365 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -901,6 +901,8 @@ libpulsecore_ at PA_MAJORMINOR@_la_SOURCES = \
 		pulsecore/remap.c pulsecore/remap.h \
 		pulsecore/remap_mmx.c pulsecore/remap_sse.c \
 		pulsecore/resampler.c pulsecore/resampler.h \
+		pulsecore/resampler/ffmpeg.c pulsecore/resampler/peaks.c \
+		pulsecore/resampler/trivial.c \
 		pulsecore/rtpoll.c pulsecore/rtpoll.h \
 		pulsecore/stream-util.c pulsecore/stream-util.h \
 		pulsecore/mix.c pulsecore/mix.h \
@@ -925,9 +927,9 @@ libpulsecore_ at PA_MAJORMINOR@_la_SOURCES = \
 		pulsecore/thread-mq.c pulsecore/thread-mq.h \
 		pulsecore/database.h
 
-libpulsecore_ at PA_MAJORMINOR@_la_CFLAGS = $(AM_CFLAGS) $(SERVER_CFLAGS) $(LIBSAMPLERATE_CFLAGS) $(LIBSPEEX_CFLAGS) $(LIBSNDFILE_CFLAGS) $(WINSOCK_CFLAGS)
+libpulsecore_ at PA_MAJORMINOR@_la_CFLAGS = $(AM_CFLAGS) $(SERVER_CFLAGS) $(LIBSNDFILE_CFLAGS) $(WINSOCK_CFLAGS)
 libpulsecore_ at PA_MAJORMINOR@_la_LDFLAGS = $(AM_LDFLAGS) -avoid-version
-libpulsecore_ at PA_MAJORMINOR@_la_LIBADD = $(AM_LIBADD) $(LIBLTDL) $(LIBSAMPLERATE_LIBS) $(LIBSPEEX_LIBS) $(LIBSNDFILE_LIBS) $(WINSOCK_LIBS) $(LTLIBICONV) libpulsecommon- at PA_MAJORMINOR@.la libpulse.la libpulsecore-foreign.la
+libpulsecore_ at PA_MAJORMINOR@_la_LIBADD = $(AM_LIBADD) $(LIBLTDL) $(LIBSNDFILE_LIBS) $(WINSOCK_LIBS) $(LTLIBICONV) libpulsecommon- at PA_MAJORMINOR@.la libpulse.la libpulsecore-foreign.la
 
 if HAVE_NEON
 noinst_LTLIBRARIES += libpulsecore_sconv_neon.la libpulsecore_mix_neon.la libpulsecore_remap_neon.la
@@ -978,6 +980,18 @@ if HAVE_SIMPLEDB
 libpulsecore_ at PA_MAJORMINOR@_la_SOURCES += pulsecore/database-simple.c
 endif
 
+if HAVE_SPEEX
+libpulsecore_ at PA_MAJORMINOR@_la_SOURCES += pulsecore/resampler/speex.c
+libpulsecore_ at PA_MAJORMINOR@_la_CFLAGS += $(LIBSPEEX_CFLAGS)
+libpulsecore_ at PA_MAJORMINOR@_la_LIBADD += $(LIBSPEEX_LIBS)
+endif
+
+if HAVE_LIBSAMPLERATE
+libpulsecore_ at PA_MAJORMINOR@_la_SOURCES += pulsecore/resampler/libsamplerate.c
+libpulsecore_ at PA_MAJORMINOR@_la_CFLAGS += $(LIBSAMPLERATE_CFLAGS)
+libpulsecore_ at PA_MAJORMINOR@_la_LIBADD += $(LIBSAMPLERATE_LIBS)
+endif
+
 # We split the foreign code off to not be annoyed by warnings we don't care about
 noinst_LTLIBRARIES += libpulsecore-foreign.la
 
diff --git a/src/pulsecore/resampler.c b/src/pulsecore/resampler.c
index 2842e65..de58f3f 100644
--- a/src/pulsecore/resampler.c
+++ b/src/pulsecore/resampler.c
@@ -25,106 +25,39 @@
 
 #include <string.h>
 
-#ifdef HAVE_LIBSAMPLERATE
-#include <samplerate.h>
-#endif
-
 #ifdef HAVE_SPEEX
 #include <speex/speex_resampler.h>
+#include <math.h>
 #endif
 
 #include <pulse/xmalloc.h>
-#include <pulsecore/sconv.h>
 #include <pulsecore/log.h>
 #include <pulsecore/macro.h>
 #include <pulsecore/strbuf.h>
-#include <pulsecore/remap.h>
 #include <pulsecore/once.h>
 #include <pulsecore/core-util.h>
-#include "ffmpeg/avcodec.h"
 
 #include "resampler.h"
 
 /* Number of samples of extra space we allow the resamplers to return */
 #define EXTRA_FRAMES 128
 
-struct pa_resampler {
-    pa_resample_method_t method;
-    pa_resample_flags_t flags;
-
-    pa_sample_spec i_ss, o_ss;
-    pa_channel_map i_cm, o_cm;
-    size_t i_fz, o_fz, w_fz, w_sz;
-    pa_mempool *mempool;
-
-    pa_memchunk to_work_format_buf;
-    pa_memchunk remap_buf;
-    pa_memchunk resample_buf;
-    pa_memchunk from_work_format_buf;
-    size_t to_work_format_buf_size;
-    size_t remap_buf_size;
-    size_t resample_buf_size;
-    size_t from_work_format_buf_size;
-
-    /* points to buffer before resampling stage, remap or to_work */
-    pa_memchunk *leftover_buf;
-    size_t *leftover_buf_size;
-
-    /* have_leftover points to leftover_in_remap or leftover_in_to_work */
-    bool *have_leftover;
-    bool leftover_in_remap;
-    bool leftover_in_to_work;
-
-    pa_sample_format_t work_format;
-    uint8_t work_channels;
-
-    pa_convert_func_t to_work_format_func;
-    pa_convert_func_t from_work_format_func;
-
-    pa_remap_t remap;
-    bool map_required;
-
-    pa_resampler_impl impl;
-};
-
-struct trivial_data { /* data specific to the trivial resampler */
-    unsigned o_counter;
-    unsigned i_counter;
-};
-
-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];
-};
-
 struct ffmpeg_data { /* data specific to ffmpeg */
     struct AVResampleContext *state;
 };
 
 static int copy_init(pa_resampler *r);
-static int trivial_init(pa_resampler*r);
-#ifdef HAVE_SPEEX
-static int speex_init(pa_resampler*r);
-#endif
-static int ffmpeg_init(pa_resampler*r);
-static int peaks_init(pa_resampler*r);
-#ifdef HAVE_LIBSAMPLERATE
-static int libsamplerate_init(pa_resampler*r);
-#endif
 
 static void setup_remap(const pa_resampler *r, pa_remap_t *m);
 static void free_remap(pa_remap_t *m);
 
-static int (* const init_table[])(pa_resampler*r) = {
+static int (* const init_table[])(pa_resampler *r) = {
 #ifdef HAVE_LIBSAMPLERATE
-    [PA_RESAMPLER_SRC_SINC_BEST_QUALITY]   = libsamplerate_init,
-    [PA_RESAMPLER_SRC_SINC_MEDIUM_QUALITY] = libsamplerate_init,
-    [PA_RESAMPLER_SRC_SINC_FASTEST]        = libsamplerate_init,
-    [PA_RESAMPLER_SRC_ZERO_ORDER_HOLD]     = libsamplerate_init,
-    [PA_RESAMPLER_SRC_LINEAR]              = libsamplerate_init,
+    [PA_RESAMPLER_SRC_SINC_BEST_QUALITY]   = pa_resampler_libsamplerate_init,
+    [PA_RESAMPLER_SRC_SINC_MEDIUM_QUALITY] = pa_resampler_libsamplerate_init,
+    [PA_RESAMPLER_SRC_SINC_FASTEST]        = pa_resampler_libsamplerate_init,
+    [PA_RESAMPLER_SRC_ZERO_ORDER_HOLD]     = pa_resampler_libsamplerate_init,
+    [PA_RESAMPLER_SRC_LINEAR]              = pa_resampler_libsamplerate_init,
 #else
     [PA_RESAMPLER_SRC_SINC_BEST_QUALITY]   = NULL,
     [PA_RESAMPLER_SRC_SINC_MEDIUM_QUALITY] = NULL,
@@ -132,30 +65,30 @@ static int (* const init_table[])(pa_resampler*r) = {
     [PA_RESAMPLER_SRC_ZERO_ORDER_HOLD]     = NULL,
     [PA_RESAMPLER_SRC_LINEAR]              = NULL,
 #endif
-    [PA_RESAMPLER_TRIVIAL]                 = trivial_init,
+    [PA_RESAMPLER_TRIVIAL]                 = pa_resampler_trivial_init,
 #ifdef HAVE_SPEEX
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+0]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+1]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+2]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+3]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+4]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+5]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+6]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+7]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+8]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+9]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FLOAT_BASE+10]     = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+0]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+1]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+2]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+3]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+4]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+5]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+6]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+7]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+8]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+9]      = speex_init,
-    [PA_RESAMPLER_SPEEX_FIXED_BASE+10]     = speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+0]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+1]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+2]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+3]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+4]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+5]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+6]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+7]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+8]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+9]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FLOAT_BASE+10]     = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+0]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+1]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+2]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+3]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+4]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+5]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+6]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+7]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+8]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+9]      = pa_resampler_speex_init,
+    [PA_RESAMPLER_SPEEX_FIXED_BASE+10]     = pa_resampler_speex_init,
 #else
     [PA_RESAMPLER_SPEEX_FLOAT_BASE+0]      = NULL,
     [PA_RESAMPLER_SPEEX_FLOAT_BASE+1]      = NULL,
@@ -180,10 +113,10 @@ static int (* const init_table[])(pa_resampler*r) = {
     [PA_RESAMPLER_SPEEX_FIXED_BASE+9]      = NULL,
     [PA_RESAMPLER_SPEEX_FIXED_BASE+10]     = NULL,
 #endif
-    [PA_RESAMPLER_FFMPEG]                  = ffmpeg_init,
+    [PA_RESAMPLER_FFMPEG]                  = pa_resampler_ffmpeg_init,
     [PA_RESAMPLER_AUTO]                    = NULL,
     [PA_RESAMPLER_COPY]                    = copy_init,
-    [PA_RESAMPLER_PEAKS]                   = peaks_init,
+    [PA_RESAMPLER_PEAKS]                   = pa_resampler_peaks_init,
 };
 
 static bool speex_is_fixed_point(void);
@@ -1408,84 +1341,6 @@ void pa_resampler_run(pa_resampler *r, const pa_memchunk *in, pa_memchunk *out)
         pa_memchunk_reset(out);
 }
 
-/*** libsamplerate based implementation ***/
-
-#ifdef HAVE_LIBSAMPLERATE
-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);
-}
-
-static int 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;
-}
-#endif
-
 /*** speex based implementation ***/
 
 static bool speex_is_fixed_point(void) {
@@ -1516,431 +1371,6 @@ static bool speex_is_fixed_point(void) {
     return result;
 }
 
-#ifdef HAVE_SPEEX
-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);
-}
-
-static int 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;
-}
-#endif
-
-/* Trivial implementation */
-
-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;
-}
-
-static int 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;
-}
-
-/* Peak finder implementation */
-
-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;
-}
-
-static int 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;
-}
-
-/*** ffmpeg based implementation ***/
-
-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);
-}
-
-static int 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)))
-        return -1;
-
-    r->impl.free = ffmpeg_free;
-    r->impl.resample = ffmpeg_resample;
-    r->impl.data = (void *) ffmpeg_data;
-
-    return 0;
-}
-
 /*** copy (noop) implementation ***/
 
 static int copy_init(pa_resampler *r) {
diff --git a/src/pulsecore/resampler.h b/src/pulsecore/resampler.h
index 058a800..13d8c35 100644
--- a/src/pulsecore/resampler.h
+++ b/src/pulsecore/resampler.h
@@ -26,10 +26,14 @@
 #include <pulse/channelmap.h>
 #include <pulsecore/memblock.h>
 #include <pulsecore/memchunk.h>
+#include <pulsecore/sconv.h>
+#include <pulsecore/remap.h>
 
 typedef struct pa_resampler pa_resampler;
 typedef struct pa_resampler_impl pa_resampler_impl;
 
+typedef struct pa_resampler pa_resampler;
+
 struct pa_resampler_impl {
     void (*free)(pa_resampler *r);
     void (*update_rates)(pa_resampler *r);
@@ -67,6 +71,45 @@ typedef enum pa_resample_flags {
     PA_RESAMPLER_NO_LFE        = 0x0008U
 } pa_resample_flags_t;
 
+struct pa_resampler {
+    pa_resample_method_t method;
+    pa_resample_flags_t flags;
+
+    pa_sample_spec i_ss, o_ss;
+    pa_channel_map i_cm, o_cm;
+    size_t i_fz, o_fz, w_fz, w_sz;
+    pa_mempool *mempool;
+
+    pa_memchunk to_work_format_buf;
+    pa_memchunk remap_buf;
+    pa_memchunk resample_buf;
+    pa_memchunk from_work_format_buf;
+    size_t to_work_format_buf_size;
+    size_t remap_buf_size;
+    size_t resample_buf_size;
+    size_t from_work_format_buf_size;
+
+    /* points to buffer before resampling stage, remap or to_work */
+    pa_memchunk *leftover_buf;
+    size_t *leftover_buf_size;
+
+    /* have_leftover points to leftover_in_remap or leftover_in_to_work */
+    bool *have_leftover;
+    bool leftover_in_remap;
+    bool leftover_in_to_work;
+
+    pa_sample_format_t work_format;
+    uint8_t work_channels;
+
+    pa_convert_func_t to_work_format_func;
+    pa_convert_func_t from_work_format_func;
+
+    pa_remap_t remap;
+    bool map_required;
+
+    pa_resampler_impl impl;
+};
+
 pa_resampler* pa_resampler_new(
         pa_mempool *pool,
         const pa_sample_spec *a,
@@ -116,4 +159,11 @@ const pa_sample_spec* pa_resampler_input_sample_spec(pa_resampler *r);
 const pa_channel_map* pa_resampler_output_channel_map(pa_resampler *r);
 const pa_sample_spec* pa_resampler_output_sample_spec(pa_resampler *r);
 
+/* Implementation specific init functions */
+int pa_resampler_ffmpeg_init(pa_resampler *r);
+int pa_resampler_libsamplerate_init(pa_resampler *r);
+int pa_resampler_peaks_init(pa_resampler *r);
+int pa_resampler_speex_init(pa_resampler *r);
+int pa_resampler_trivial_init(pa_resampler*r);
+
 #endif
diff --git a/src/pulsecore/resampler/ffmpeg.c b/src/pulsecore/resampler/ffmpeg.c
new file mode 100644
index 0000000..9e35236
--- /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, write to the Free Software
+  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+  USA.
+***/
+
+#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)))
+        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..48ea594
--- /dev/null
+++ b/src/pulsecore/resampler/libsamplerate.c
@@ -0,0 +1,102 @@
+/***
+  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, write to the Free Software
+  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+  USA.
+***/
+
+#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..ef9e99e
--- /dev/null
+++ b/src/pulsecore/resampler/peaks.c
@@ -0,0 +1,163 @@
+/***
+  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, write to the Free Software
+  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+  USA.
+***/
+
+#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/speex.c b/src/pulsecore/resampler/speex.c
new file mode 100644
index 0000000..faeef76
--- /dev/null
+++ b/src/pulsecore/resampler/speex.c
@@ -0,0 +1,151 @@
+/***
+  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, write to the Free Software
+  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+  USA.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <speex/speex_resampler.h>
+
+#include "pulsecore/resampler.h"
+
+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..09c81ee
--- /dev/null
+++ b/src/pulsecore/resampler/trivial.c
@@ -0,0 +1,102 @@
+/***
+  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, write to the Free Software
+  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+  USA.
+***/
+
+#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;
+}
-- 
1.9.1



More information about the pulseaudio-discuss mailing list