1 files changed, 34 insertions, 458 deletions

diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
index c18f8cc28a..55eb6b0f20 100644
--- a/audio/filter/af_lavrresample.c
+++ b/audio/filter/af_lavrresample.c
@@ -27,326 +27,24 @@
 #include <math.h>
 #include <assert.h>
 
-#include <libavutil/opt.h>
-#include <libavutil/common.h>
-#include <libavutil/samplefmt.h>
-#include <libavutil/channel_layout.h>
-#include <libavutil/mathematics.h>
-
 #include "common/common.h"
 #include "config.h"
 
-#define HAVE_LIBSWRESAMPLE HAVE_IS_FFMPEG
-#define HAVE_LIBAVRESAMPLE HAVE_IS_LIBAV
-
-#if HAVE_LIBAVRESAMPLE
-#include <libavresample/avresample.h>
-#elif HAVE_LIBSWRESAMPLE
-#include <libswresample/swresample.h>
-#define AVAudioResampleContext SwrContext
-#define avresample_alloc_context swr_alloc
-#define avresample_open swr_init
-#define avresample_close(x) do { } while(0)
-#define avresample_free swr_free
-#define avresample_available(x) 0
-#define avresample_convert(ctx, out, out_planesize, out_samples, in, in_planesize, in_samples) \
-    swr_convert(ctx, out, out_samples, (const uint8_t**)(in), in_samples)
-#define avresample_set_channel_mapping swr_set_channel_mapping
-#define avresample_set_compensation swr_set_compensation
-#else
-#error "config.h broken or no resampler found"
-#endif
-
 #include "common/av_common.h"
 #include "common/msg.h"
 #include "options/m_option.h"
 #include "audio/filter/af.h"
 #include "audio/fmt-conversion.h"
 #include "osdep/endian.h"
-
-struct af_resample_opts {
-    int filter_size;
-    int phase_shift;
-    int linear;
-    double cutoff;
-    int normalize;
-};
+#include "audio/aconverter.h"
 
 struct af_resample {
     int allow_detach;
-    char **avopts;
     double playback_speed;
-    bool is_resampling;
-    struct AVAudioResampleContext *avrctx;
-    struct mp_audio avrctx_fmt; // output format of avrctx
-    struct mp_audio pool_fmt; // format used to allocate frames for avrctx output
-    struct mp_audio pre_out_fmt; // format before final conversion (S24)
-    struct AVAudioResampleContext *avrctx_out; // for output channel reordering
-    struct af_resample_opts opts;  // opts requested by the user
-    // At least libswresample keeps a pointer around for this:
-    int reorder_in[MP_NUM_CHANNELS];
-    int reorder_out[MP_NUM_CHANNELS];
-    struct mp_audio_pool *reorder_buffer;
-
-    int in_rate_af; // filter input sample rate
-    int in_rate;    // actual rate (used by lavr), adjusted for playback speed
-    int in_format;
-    struct mp_chmap in_channels;
-    int out_rate;
-    int out_format;
-    struct mp_chmap out_channels;
-};
-
-#if HAVE_LIBAVRESAMPLE
-static double get_delay(struct af_resample *s)
-{
-    return avresample_get_delay(s->avrctx) / (double)s->in_rate +
-           avresample_available(s->avrctx) / (double)s->out_rate;
-}
-static int get_out_samples(struct af_resample *s, int in_samples)
-{
-    return avresample_get_out_samples(s->avrctx, in_samples);
-}
-#else
-static double get_delay(struct af_resample *s)
-{
-    int64_t base = s->in_rate * (int64_t)s->out_rate;
-    return swr_get_delay(s->avrctx, base) / (double)base;
-}
-static int get_out_samples(struct af_resample *s, int in_samples)
-{
-    return swr_get_out_samples(s->avrctx, in_samples);
-}
-#endif
-
-static void close_lavrr(struct af_instance *af)
-{
-    struct af_resample *s = af->priv;
-
-    if (s->avrctx)
-        avresample_close(s->avrctx);
-    avresample_free(&s->avrctx);
-    if (s->avrctx_out)
-        avresample_close(s->avrctx_out);
-    avresample_free(&s->avrctx_out);
-}
-
-static int resample_frame(struct AVAudioResampleContext *r,
-                          struct mp_audio *out, struct mp_audio *in)
-{
-    return avresample_convert(r,
-        out ? (uint8_t **)out->planes : NULL,
-        out ? mp_audio_get_allocated_size(out) : 0,
-        out ? out->samples : 0,
-        in ? (uint8_t **)in->planes : NULL,
-        in ? mp_audio_get_allocated_size(in) : 0,
-        in ? in->samples : 0);
-}
-
-static double af_resample_default_cutoff(int filter_size)
-{
-    return FFMAX(1.0 - 6.5 / (filter_size + 8), 0.80);
-}
-
-static int rate_from_speed(int rate, double speed)
-{
-    return lrint(rate * speed);
-}
-
-static struct mp_chmap fudge_pairs[][2] = {
-    {MP_CHMAP2(BL,  BR),  MP_CHMAP2(SL,  SR)},
-    {MP_CHMAP2(SL,  SR),  MP_CHMAP2(BL,  BR)},
-    {MP_CHMAP2(SDL, SDR), MP_CHMAP2(SL,  SR)},
-    {MP_CHMAP2(SL,  SR),  MP_CHMAP2(SDL, SDR)},
+    struct mp_resample_opts opts;
+    struct mp_aconverter *converter;
 };
 
-// Modify out_layout and return the new value. The intention is reducing the
-// loss libswresample's rematrixing will cause by exchanging similar, but
-// strictly speaking incompatible channel pairs. For example, 7.1 should be
-// changed to 7.1(wide) without dropping the SL/SR channels. (We still leave
-// it to libswresample to create the remix matrix.)
-static uint64_t fudge_layout_conversion(struct af_instance *af,
-                                        uint64_t in, uint64_t out)
-{
-    for (int n = 0; n < MP_ARRAY_SIZE(fudge_pairs); n++) {
-        uint64_t a = mp_chmap_to_lavc(&fudge_pairs[n][0]);
-        uint64_t b = mp_chmap_to_lavc(&fudge_pairs[n][1]);
-        if ((in & a) == a && (in & b) == 0 &&
-            (out & a) == 0 && (out & b) == b)
-        {
-            out = (out & ~b) | a;
-
-            MP_VERBOSE(af, "Fudge: %s -> %s\n",
-                       mp_chmap_to_str(&fudge_pairs[n][0]),
-                       mp_chmap_to_str(&fudge_pairs[n][1]));
-        }
-    }
-    return out;
-}
-
-// mp_chmap_get_reorder() performs:
-//  to->speaker[n] = from->speaker[src[n]]
-// but libavresample does:
-//  to->speaker[dst[n]] = from->speaker[n]
-static void transpose_order(int *map, int num)
-{
-    int nmap[MP_NUM_CHANNELS] = {0};
-    for (int n = 0; n < num; n++) {
-        for (int i = 0; i < num; i++) {
-            if (map[n] == i)
-                nmap[i] = n;
-        }
-    }
-    memcpy(map, nmap, sizeof(nmap));
-}
-
-static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
-                           struct mp_audio *out, bool verbose)
-{
-    struct af_resample *s = af->priv;
-
-    close_lavrr(af);
-
-    s->avrctx = avresample_alloc_context();
-    s->avrctx_out = avresample_alloc_context();
-    if (!s->avrctx || !s->avrctx_out)
-        goto error;
-
-    enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
-    enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
-    enum AVSampleFormat out_samplefmtp = av_get_planar_sample_fmt(out_samplefmt);
-
-    if (in_samplefmt == AV_SAMPLE_FMT_NONE ||
-        out_samplefmt == AV_SAMPLE_FMT_NONE ||
-        out_samplefmtp == AV_SAMPLE_FMT_NONE)
-        goto error;
-
-    s->out_rate    = out->rate;
-    s->in_rate_af  = in->rate;
-    s->in_rate     = rate_from_speed(in->rate, s->playback_speed);
-    s->out_format  = out->format;
-    s->in_format   = in->format;
-    s->out_channels= out->channels;
-    s->in_channels = in->channels;
-
-    av_opt_set_int(s->avrctx, "filter_size",        s->opts.filter_size, 0);
-    av_opt_set_int(s->avrctx, "phase_shift",        s->opts.phase_shift, 0);
-    av_opt_set_int(s->avrctx, "linear_interp",      s->opts.linear, 0);
-
-    av_opt_set_double(s->avrctx, "cutoff",          s->opts.cutoff, 0);
-
-    int normalize = s->opts.normalize;
-    if (normalize < 0)
-        normalize = af->opts->audio_normalize;
-#if HAVE_LIBSWRESAMPLE
-    av_opt_set_double(s->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0);
-#else
-    av_opt_set_int(s->avrctx, "normalize_mix_level", !!normalize, 0);
-#endif
-
-    if (mp_set_avopts(af->log, s->avrctx, s->avopts) < 0)
-        goto error;
-
-    struct mp_chmap map_in = in->channels;
-    struct mp_chmap map_out = out->channels;
-
-    // Try not to do any remixing if at least one is "unknown".
-    if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&map_out)) {
-        mp_chmap_set_unknown(&map_in, map_in.num);
-        mp_chmap_set_unknown(&map_out, map_out.num);
-    }
-
-    // unchecked: don't take any channel reordering into account
-    uint64_t in_ch_layout = mp_chmap_to_lavc_unchecked(&map_in);
-    uint64_t out_ch_layout = mp_chmap_to_lavc_unchecked(&map_out);
-
-    struct mp_chmap in_lavc, out_lavc;
-    mp_chmap_from_lavc(&in_lavc, in_ch_layout);
-    mp_chmap_from_lavc(&out_lavc, out_ch_layout);
-
-    if (verbose && !mp_chmap_equals(&in_lavc, &out_lavc)) {
-        MP_VERBOSE(af, "Remix: %s -> %s\n", mp_chmap_to_str(&in_lavc),
-                                            mp_chmap_to_str(&out_lavc));
-    }
-
-    if (in_lavc.num != map_in.num) {
-        // For handling NA channels, we would have to add a planarization step.
-        MP_FATAL(af, "Unsupported channel remapping.\n");
-        goto error;
-    }
-
-    mp_chmap_get_reorder(s->reorder_in, &map_in, &in_lavc);
-    transpose_order(s->reorder_in, map_in.num);
-
-    if (mp_chmap_equals(&out_lavc, &map_out)) {
-        // No intermediate step required - output new format directly.
-        out_samplefmtp = out_samplefmt;
-    } else {
-        // Verify that we really just reorder and/or insert NA channels.
-        struct mp_chmap withna = out_lavc;
-        mp_chmap_fill_na(&withna, map_out.num);
-        if (withna.num != map_out.num)
-            goto error;
-    }
-    mp_chmap_get_reorder(s->reorder_out, &out_lavc, &map_out);
-
-    s->avrctx_fmt = *out;
-    mp_audio_set_channels(&s->avrctx_fmt, &out_lavc);
-    mp_audio_set_format(&s->avrctx_fmt, af_from_avformat(out_samplefmtp));
-
-    s->pre_out_fmt = *out;
-
-    // If there are NA channels, the final output will have more channels than
-    // the avrctx output. Also, avrctx will output planar (out_samplefmtp was
-    // not overwritten). Allocate the output frame with more channels, so the
-    // NA channels can be trivially added.
-    s->pool_fmt = s->avrctx_fmt;
-    if (map_out.num > out_lavc.num)
-        mp_audio_set_channels(&s->pool_fmt, &map_out);
-
-    out_ch_layout = fudge_layout_conversion(af, in_ch_layout, out_ch_layout);
-
-    // Real conversion; output is input to avrctx_out.
-    av_opt_set_int(s->avrctx, "in_channel_layout",  in_ch_layout, 0);
-    av_opt_set_int(s->avrctx, "out_channel_layout", out_ch_layout, 0);
-    av_opt_set_int(s->avrctx, "in_sample_rate",     s->in_rate, 0);
-    av_opt_set_int(s->avrctx, "out_sample_rate",    s->out_rate, 0);
-    av_opt_set_int(s->avrctx, "in_sample_fmt",      in_samplefmt, 0);
-    av_opt_set_int(s->avrctx, "out_sample_fmt",     out_samplefmtp, 0);
-
-    // Just needs the correct number of channels for deplanarization.
-    struct mp_chmap fake_chmap;
-    mp_chmap_set_unknown(&fake_chmap, map_out.num);
-    uint64_t fake_out_ch_layout = mp_chmap_to_lavc_unchecked(&fake_chmap);
-    if (!fake_out_ch_layout)
-        goto error;
-    av_opt_set_int(s->avrctx_out, "in_channel_layout",  fake_out_ch_layout, 0);
-    av_opt_set_int(s->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
-
-    av_opt_set_int(s->avrctx_out, "in_sample_fmt",      out_samplefmtp, 0);
-    av_opt_set_int(s->avrctx_out, "out_sample_fmt",     out_samplefmt, 0);
-    av_opt_set_int(s->avrctx_out, "in_sample_rate",     s->out_rate, 0);
-    av_opt_set_int(s->avrctx_out, "out_sample_rate",    s->out_rate, 0);
-
-    // API has weird requirements, quoting avresample.h:
-    //  * This function can only be called when the allocated context is not open.
-    //  * Also, the input channel layout must have already been set.
-    avresample_set_channel_mapping(s->avrctx, s->reorder_in);
-
-    s->is_resampling = false;
-
-    if (avresample_open(s->avrctx) < 0 || avresample_open(s->avrctx_out) < 0) {
-        MP_ERR(af, "Cannot open Libavresample Context. \n");
-        goto error;
-    }
-    return AF_OK;
-
-error:
-    close_lavrr(af);
-    return AF_ERROR;
-}
-
-
 static int control(struct af_instance *af, int cmd, void *arg)
 {
     struct af_resample *s = af->priv;
@@ -378,8 +76,12 @@ static int control(struct af_instance *af, int cmd, void *arg)
                 mp_chmap_equals(&in->channels, &orig_in.channels))
                 ? AF_OK : AF_FALSE;
 
-        if (r == AF_OK)
-            r = configure_lavrr(af, in, out, true);
+        if (r == AF_OK) {
+            if (!mp_aconverter_reconfig(s->converter,
+                    in->rate, in->format, in->channels,
+                    out->rate, out->format, out->channels))
+                r = AF_ERROR;
+        }
         return r;
     }
     case AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE: {
@@ -387,17 +89,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
         return AF_OK;
     }
     case AF_CONTROL_RESET:
-        if (s->avrctx) {
-#if HAVE_LIBSWRESAMPLE
-            swr_close(s->avrctx);
-            if (swr_init(s->avrctx) < 0) {
-                close_lavrr(af);
-                return AF_ERROR;
-            }
-#else
-            while (avresample_read(s->avrctx, NULL, 1000) > 0) {}
-#endif
-        }
+        mp_aconverter_flush(s->converter);
         return AF_OK;
     }
     return AF_UNKNOWN;
@@ -405,149 +97,40 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
 static void uninit(struct af_instance *af)
 {
-    close_lavrr(af);
-}
-
-// The LSB is always ignored.
-#if BYTE_ORDER == BIG_ENDIAN
-#define SHIFT24(x) ((3-(x))*8)
-#else
-#define SHIFT24(x) (((x)+1)*8)
-#endif
-
-static void extra_output_conversion(struct af_instance *af, struct mp_audio *mpa)
-{
-    for (int p = 0; p < mpa->num_planes; p++) {
-        void *ptr = mpa->planes[p];
-        int total = mpa->samples * mpa->spf;
-        if (af_fmt_from_planar(mpa->format) == AF_FORMAT_FLOAT) {
-            for (int s = 0; s < total; s++)
-                ((float *)ptr)[s] = av_clipf(((float *)ptr)[s], -1.0f, 1.0f);
-        } else if (af_fmt_from_planar(mpa->format) == AF_FORMAT_DOUBLE) {
-            for (int s = 0; s < total; s++)
-                ((double *)ptr)[s] = MPCLAMP(((double *)ptr)[s], -1.0, 1.0);
-        }
-    }
-}
-
-// This relies on the tricky way mpa was allocated.
-static void reorder_planes(struct mp_audio *mpa, int *reorder,
-                           struct mp_chmap *newmap)
-{
-    struct mp_audio prev = *mpa;
-    mp_audio_set_channels(mpa, newmap);
-
-    // The trailing planes were never written by avrctx, they're the NA channels.
-    int next_na = prev.num_planes;
+    struct af_resample *s = af->priv;
 
-    for (int n = 0; n < mpa->num_planes; n++) {
-        int src = reorder[n];
-        assert(src >= -1 && src < prev.num_planes);
-        if (src >= 0) {
-            mpa->planes[n] = prev.planes[src];
-        } else {
-            assert(next_na < mpa->num_planes);
-            mpa->planes[n] = prev.planes[next_na++];
-            af_fill_silence(mpa->planes[n], mpa->sstride * mpa->samples,
-                            mpa->format);
-        }
-    }
+    talloc_free(s->converter);
 }
 
-static int filter_resample(struct af_instance *af, struct mp_audio *in)
+static int filter(struct af_instance *af, struct mp_audio *in)
 {
     struct af_resample *s = af->priv;
-    struct mp_audio *out = NULL;
-
-    if (!s->avrctx)
-        goto error;
 
-    int samples = get_out_samples(s, in ? in->samples : 0);
-
-    struct mp_audio out_format = s->pool_fmt;
-    out = mp_audio_pool_get(af->out_pool, &out_format, samples);
-    if (!out)
-        goto error;
-    if (in)
-        mp_audio_copy_attributes(out, in);
-
-    if (out->samples) {
-        out->samples = resample_frame(s->avrctx, out, in);
-        if (out->samples < 0)
-            goto error;
-    }
-
-    struct mp_audio real_out = *out;
-    mp_audio_copy_config(out, &s->avrctx_fmt);
-
-    if (out->samples && !mp_audio_config_equals(out, &s->pre_out_fmt)) {
-        assert(af_fmt_is_planar(out->format) && out->format == real_out.format);
-        reorder_planes(out, s->reorder_out, &s->pool_fmt.channels);
-        if (!mp_audio_config_equals(out, &s->pre_out_fmt)) {
-            struct mp_audio *new = mp_audio_pool_get(s->reorder_buffer,
-                                                     &s->pre_out_fmt,
-                                                     out->samples);
-            if (!new)
-                goto error;
-            mp_audio_copy_attributes(new, out);
-            int out_samples = resample_frame(s->avrctx_out, new, out);
-            talloc_free(out);
-            out = new;
-            if (out_samples != new->samples)
-                goto error;
-        }
-    }
+    mp_aconverter_set_speed(s->converter, s->playback_speed);
 
-    extra_output_conversion(af, out);
+    af->filter_out(af);
 
+    struct mp_aframe *aframe = mp_audio_to_aframe(in);
+    if (!aframe && in)
+        return -1;
     talloc_free(in);
-    if (out->samples) {
-        af_add_output_frame(af, out);
-    } else {
-        talloc_free(out);
-    }
-
-    af->delay = get_delay(s);
+    bool ok = mp_aconverter_write_input(s->converter, aframe);
+    if (!ok)
+        talloc_free(aframe);
 
-    return 0;
-error:
-    talloc_free(in);
-    talloc_free(out);
-    return -1;
+    return ok ? 0 : -1;
 }
 
-static int filter(struct af_instance *af, struct mp_audio *in)
+static int filter_out(struct af_instance *af)
 {
     struct af_resample *s = af->priv;
-
-    int new_rate = rate_from_speed(s->in_rate_af, s->playback_speed);
-    if (s->avrctx && !(!s->is_resampling && new_rate == s->in_rate)) {
-        AVRational r = av_d2q(s->playback_speed * s->in_rate_af / s->in_rate,
-                              INT_MAX / 2);
-        // Essentially, swr/avresample_set_compensation() does 2 things:
-        // - adjust output sample rate by sample_delta/compensation_distance
-        // - reset the adjustment after compensation_distance output samples
-        // Increase the compensation_distance to avoid undesired reset
-        // semantics - we want to keep the ratio for the whole frame we're
-        // feeding it, until the next filter() call.
-        int mult = INT_MAX / 2 / MPMAX(MPMAX(abs(r.num), abs(r.den)), 1);
-        r = (AVRational){ r.num * mult, r.den * mult };
-        if (avresample_set_compensation(s->avrctx, r.den - r.num, r.den) >= 0) {
-            new_rate = s->in_rate;
-            s->is_resampling = true;
-        }
-    }
-
-    bool need_reinit = fabs(new_rate / (double)s->in_rate - 1) > 0.01;
-    if (need_reinit && new_rate != s->in_rate) {
-        // Before reconfiguring, drain the audio that is still buffered
-        // in the resampler.
-        filter_resample(af, NULL);
-        // Reinitialize resampler.
-        configure_lavrr(af, &af->fmt_in, &af->fmt_out, false);
-    }
-
-    return filter_resample(af, in);
+    bool eof;
+    struct mp_aframe *out = mp_aconverter_read_output(s->converter, &eof);
+    if (out)
+        af_add_output_frame(af, mp_audio_from_aframe(out));
+    talloc_free(out);
+    af->delay = mp_aconverter_get_latency(s->converter);
+    return 0;
 }
 
 static int af_open(struct af_instance *af)
@@ -557,11 +140,9 @@ static int af_open(struct af_instance *af)
     af->control = control;
     af->uninit  = uninit;
     af->filter_frame = filter;
+    af->filter_out = filter_out;
 
-    if (s->opts.cutoff <= 0.0)
-        s->opts.cutoff = af_resample_default_cutoff(s->opts.filter_size);
-
-    s->reorder_buffer = mp_audio_pool_create(s);
+    s->converter = mp_aconverter_create(af->global, af->log, &s->opts);
 
     return AF_OK;
 }
@@ -574,12 +155,7 @@ const struct af_info af_info_lavrresample = {
     .open = af_open,
     .priv_size = sizeof(struct af_resample),
     .priv_defaults = &(const struct af_resample) {
-        .opts = {
-            .filter_size = 16,
-            .cutoff      = 0.0,
-            .phase_shift = 10,
-            .normalize   = -1,
-        },
+        .opts = MP_RESAMPLE_OPTS_DEF,
         .playback_speed = 1.0,
         .allow_detach = 1,
     },
@@ -591,7 +167,7 @@ const struct af_info af_info_lavrresample = {
         OPT_FLAG("detach", allow_detach, 0),
         OPT_CHOICE("normalize", opts.normalize, 0,
                    ({"no", 0}, {"yes", 1}, {"auto", -1})),
-        OPT_KEYVALUELIST("o", avopts, 0),
+        OPT_KEYVALUELIST("o", opts.avopts, 0),
         {0}
     },
 };