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authorwm4 <wm4@nowhere>2017-09-21 11:07:02 +0200
committerwm4 <wm4@nowhere>2017-09-21 12:42:09 +0200
commit3a2d5e68acb2ac0f8b09b896907a692b1c48c6b3 (patch)
tree13a7e321ce6accceca720fb654cdf7ea8eee0afc /audio/filter/af_lavrresample.c
parentcaaa1189ba75a7df9a4d02f7747d6c0bf3b05012 (diff)
downloadmpv-3a2d5e68acb2ac0f8b09b896907a692b1c48c6b3.tar.bz2
mpv-3a2d5e68acb2ac0f8b09b896907a692b1c48c6b3.tar.xz
audio: move libswresample wrapper out of audio filter code
Move it from af_lavrresample.c to a new aconverter.c file, which is independent from the filter chain code. It also doesn't use mp_audio, and thus has no GPL dependencies. Preparation for later commits. Not particularly well tested, so have fun.
Diffstat (limited to 'audio/filter/af_lavrresample.c')
-rw-r--r--audio/filter/af_lavrresample.c492
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}
},
};