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mpv/audio/aconverter.c

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/*
* This file is part of mpv.
*
* mpv 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.
*
* mpv 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <libavutil/opt.h>
#include <libavutil/common.h>
#include <libavutil/samplefmt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/mathematics.h>
#include "config.h"
#include "common/common.h"
#include "common/av_common.h"
#include "common/msg.h"
#include "options/m_config.h"
#include "options/m_option.h"
#include "aconverter.h"
#include "aframe.h"
#include "fmt-conversion.h"
#include "format.h"
#define HAVE_LIBSWRESAMPLE HAVE_FFMPEG_MPV
#define HAVE_LIBAVRESAMPLE HAVE_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
struct mp_aconverter {
struct mp_log *log;
struct mpv_global *global;
double playback_speed;
bool is_resampling;
bool passthrough_mode;
struct AVAudioResampleContext *avrctx;
struct mp_aframe *avrctx_fmt; // output format of avrctx
struct mp_aframe *pool_fmt; // format used to allocate frames for avrctx output
struct mp_aframe *pre_out_fmt; // format before final conversion
struct AVAudioResampleContext *avrctx_out; // for output channel reordering
const struct mp_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_aframe_pool *reorder_buffer;
struct mp_aframe_pool *out_pool;
int in_rate_user; // user 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;
struct mp_aframe *input; // queued input frame
bool input_eof; // queued input EOF
struct mp_aframe *output; // queued output frame
bool output_eof; // queued output EOF
};
#if HAVE_LIBAVRESAMPLE
static double get_delay(struct mp_aconverter *p)
{
return avresample_get_delay(p->avrctx) / (double)p->in_rate +
avresample_available(p->avrctx) / (double)p->out_rate;
}
static int get_out_samples(struct mp_aconverter *p, int in_samples)
{
return avresample_get_out_samples(p->avrctx, in_samples);
}
#else
static double get_delay(struct mp_aconverter *p)
{
int64_t base = p->in_rate * (int64_t)p->out_rate;
return swr_get_delay(p->avrctx, base) / (double)base;
}
static int get_out_samples(struct mp_aconverter *p, int in_samples)
{
return swr_get_out_samples(p->avrctx, in_samples);
}
#endif
static void close_lavrr(struct mp_aconverter *p)
{
if (p->avrctx)
avresample_close(p->avrctx);
avresample_free(&p->avrctx);
if (p->avrctx_out)
avresample_close(p->avrctx_out);
avresample_free(&p->avrctx_out);
TA_FREEP(&p->pre_out_fmt);
TA_FREEP(&p->avrctx_fmt);
TA_FREEP(&p->pool_fmt);
}
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)},
};
// 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 mp_aconverter *p,
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(p, "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 bool configure_lavrr(struct mp_aconverter *p, bool verbose)
{
close_lavrr(p);
p->in_rate = rate_from_speed(p->in_rate_user, p->playback_speed);
p->passthrough_mode = p->opts->allow_passthrough &&
p->in_rate == p->out_rate &&
p->in_format == p->out_format &&
mp_chmap_equals(&p->in_channels, &p->out_channels);
if (p->passthrough_mode)
return true;
p->avrctx = avresample_alloc_context();
p->avrctx_out = avresample_alloc_context();
if (!p->avrctx || !p->avrctx_out)
goto error;
enum AVSampleFormat in_samplefmt = af_to_avformat(p->in_format);
enum AVSampleFormat out_samplefmt = af_to_avformat(p->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;
av_opt_set_int(p->avrctx, "filter_size", p->opts->filter_size, 0);
av_opt_set_int(p->avrctx, "phase_shift", p->opts->phase_shift, 0);
av_opt_set_int(p->avrctx, "linear_interp", p->opts->linear, 0);
double cutoff = p->opts->cutoff;
if (cutoff <= 0.0)
cutoff = MPMAX(1.0 - 6.5 / (p->opts->filter_size + 8), 0.80);
av_opt_set_double(p->avrctx, "cutoff", cutoff, 0);
int global_normalize;
mp_read_option_raw(p->global, "audio-normalize-downmix", &m_option_type_flag,
&global_normalize);
int normalize = p->opts->normalize;
if (normalize < 0)
normalize = global_normalize;
#if HAVE_LIBSWRESAMPLE
av_opt_set_double(p->avrctx, "rematrix_maxval", normalize ? 1 : 1000, 0);
#else
av_opt_set_int(p->avrctx, "normalize_mix_level", !!normalize, 0);
#endif
if (mp_set_avopts(p->log, p->avrctx, p->opts->avopts) < 0)
goto error;
struct mp_chmap map_in = p->in_channels;
struct mp_chmap map_out = p->out_channels;
// Try not to do any remixing if at least one is "unknown". Some corner
// cases also benefit from disabling all channel handling logic if the
// src/dst layouts are the same (like fl-fr-na -> fl-fr-na).
if (mp_chmap_is_unknown(&map_in) || mp_chmap_is_unknown(&map_out) ||
mp_chmap_equals(&map_in, &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(p, "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(p, "Unsupported input channel layout %s.\n",
mp_chmap_to_str(&map_in));
goto error;
}
mp_chmap_get_reorder(p->reorder_in, &map_in, &in_lavc);
transpose_order(p->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(p->reorder_out, &out_lavc, &map_out);
p->pre_out_fmt = mp_aframe_create();
mp_aframe_set_rate(p->pre_out_fmt, p->out_rate);
mp_aframe_set_chmap(p->pre_out_fmt, &p->out_channels);
mp_aframe_set_format(p->pre_out_fmt, p->out_format);
p->avrctx_fmt = mp_aframe_create();
mp_aframe_config_copy(p->avrctx_fmt, p->pre_out_fmt);
mp_aframe_set_chmap(p->avrctx_fmt, &out_lavc);
mp_aframe_set_format(p->avrctx_fmt, af_from_avformat(out_samplefmtp));
// 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.
p->pool_fmt = mp_aframe_create();
mp_aframe_config_copy(p->pool_fmt, p->avrctx_fmt);
if (map_out.num > out_lavc.num)
mp_aframe_set_chmap(p->pool_fmt, &map_out);
out_ch_layout = fudge_layout_conversion(p, in_ch_layout, out_ch_layout);
// Real conversion; output is input to avrctx_out.
av_opt_set_int(p->avrctx, "in_channel_layout", in_ch_layout, 0);
av_opt_set_int(p->avrctx, "out_channel_layout", out_ch_layout, 0);
av_opt_set_int(p->avrctx, "in_sample_rate", p->in_rate, 0);
av_opt_set_int(p->avrctx, "out_sample_rate", p->out_rate, 0);
av_opt_set_int(p->avrctx, "in_sample_fmt", in_samplefmt, 0);
av_opt_set_int(p->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(p->avrctx_out, "in_channel_layout", fake_out_ch_layout, 0);
av_opt_set_int(p->avrctx_out, "out_channel_layout", fake_out_ch_layout, 0);
av_opt_set_int(p->avrctx_out, "in_sample_fmt", out_samplefmtp, 0);
av_opt_set_int(p->avrctx_out, "out_sample_fmt", out_samplefmt, 0);
av_opt_set_int(p->avrctx_out, "in_sample_rate", p->out_rate, 0);
av_opt_set_int(p->avrctx_out, "out_sample_rate", p->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(p->avrctx, p->reorder_in);
p->is_resampling = false;
if (avresample_open(p->avrctx) < 0 || avresample_open(p->avrctx_out) < 0) {
MP_ERR(p, "Cannot open Libavresample Context. \n");
goto error;
}
return true;
error:
close_lavrr(p);
return false;
}
bool mp_aconverter_reconfig(struct mp_aconverter *p,
int in_rate, int in_format, struct mp_chmap in_channels,
int out_rate, int out_format, struct mp_chmap out_channels)
{
close_lavrr(p);
TA_FREEP(&p->input);
TA_FREEP(&p->output);
p->input_eof = p->output_eof = false;
p->playback_speed = 1.0;
p->in_rate_user = in_rate;
p->in_format = in_format;
p->in_channels = in_channels;
p->out_rate = out_rate;
p->out_format = out_format;
p->out_channels = out_channels;
return configure_lavrr(p, true);
}
void mp_aconverter_flush(struct mp_aconverter *p)
{
if (!p->avrctx)
return;
#if HAVE_LIBSWRESAMPLE
swr_close(p->avrctx);
if (swr_init(p->avrctx) < 0)
close_lavrr(p);
#else
while (avresample_read(p->avrctx, NULL, 1000) > 0) {}
#endif
}
void mp_aconverter_set_speed(struct mp_aconverter *p, double speed)
{
p->playback_speed = speed;
}
static void extra_output_conversion(struct mp_aframe *mpa)
{
int format = af_fmt_from_planar(mp_aframe_get_format(mpa));
int num_planes = mp_aframe_get_planes(mpa);
uint8_t **planes = mp_aframe_get_data_rw(mpa);
if (!planes)
return;
for (int p = 0; p < num_planes; p++) {
void *ptr = planes[p];
int total = mp_aframe_get_total_plane_samples(mpa);
if (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 (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 bool reorder_planes(struct mp_aframe *mpa, int *reorder,
struct mp_chmap *newmap)
{
if (!mp_aframe_set_chmap(mpa, newmap))
return false;
int num_planes = newmap->num;
uint8_t **planes = mp_aframe_get_data_rw(mpa);
uint8_t *old_planes[MP_NUM_CHANNELS];
assert(num_planes <= MP_NUM_CHANNELS);
for (int n = 0; n < num_planes; n++)
old_planes[n] = planes[n];
int next_na = 0;
for (int n = 0; n < num_planes; n++)
next_na += newmap->speaker[n] != MP_SPEAKER_ID_NA;
for (int n = 0; n < num_planes; n++) {
int src = reorder[n];
assert(src >= -1 && src < num_planes);
if (src >= 0) {
planes[n] = old_planes[src];
} else {
assert(next_na < num_planes);
planes[n] = old_planes[next_na++];
// The NA planes were never written by avrctx, so clear them.
af_fill_silence(planes[n],
mp_aframe_get_sstride(mpa) * mp_aframe_get_size(mpa),
mp_aframe_get_format(mpa));
}
}
return true;
}
static int resample_frame(struct AVAudioResampleContext *r,
struct mp_aframe *out, struct mp_aframe *in)
{
// Be aware that the channel layout and count can be different for in and
// out frames. In some situations the caller will fix up the frames before
// or after conversion. The sample rates can also be different.
AVFrame *av_i = in ? mp_aframe_get_raw_avframe(in) : NULL;
AVFrame *av_o = out ? mp_aframe_get_raw_avframe(out) : NULL;
return avresample_convert(r,
av_o ? av_o->extended_data : NULL,
av_o ? av_o->linesize[0] : 0,
av_o ? av_o->nb_samples : 0,
av_i ? av_i->extended_data : NULL,
av_i ? av_i->linesize[0] : 0,
av_i ? av_i->nb_samples : 0);
}
static void filter_resample(struct mp_aconverter *p, struct mp_aframe *in)
{
struct mp_aframe *out = NULL;
if (!p->avrctx)
goto error;
int samples = get_out_samples(p, in ? mp_aframe_get_size(in) : 0);
out = mp_aframe_create();
mp_aframe_config_copy(out, p->pool_fmt);
if (mp_aframe_pool_allocate(p->out_pool, out, samples) < 0)
goto error;
int out_samples = 0;
if (samples) {
out_samples = resample_frame(p->avrctx, out, in);
if (out_samples < 0 || out_samples > samples)
goto error;
mp_aframe_set_size(out, out_samples);
}
struct mp_chmap out_chmap;
if (!mp_aframe_get_chmap(p->pool_fmt, &out_chmap))
goto error;
if (!reorder_planes(out, p->reorder_out, &out_chmap))
goto error;
if (!mp_aframe_config_equals(out, p->pre_out_fmt)) {
struct mp_aframe *new = mp_aframe_create();
mp_aframe_config_copy(new, p->pre_out_fmt);
if (mp_aframe_pool_allocate(p->reorder_buffer, new, out_samples) < 0) {
talloc_free(new);
goto error;
}
int got = 0;
if (out_samples)
got = resample_frame(p->avrctx_out, new, out);
talloc_free(out);
out = new;
if (got != out_samples)
goto error;
}
extra_output_conversion(out);
if (in)
mp_aframe_copy_attributes(out, in);
if (out_samples) {
p->output = out;
} else {
talloc_free(out);
}
p->output_eof = !in; // we've read everything
return;
error:
talloc_free(out);
MP_ERR(p, "Error on resampling.\n");
}
static void filter(struct mp_aconverter *p)
{
if (p->output || p->output_eof || !(p->input || p->input_eof))
return;
int new_rate = rate_from_speed(p->in_rate_user, p->playback_speed);
if (p->passthrough_mode && new_rate != p->in_rate)
configure_lavrr(p, false);
if (p->passthrough_mode) {
p->output = p->input;
p->input = NULL;
p->output_eof = p->input_eof;
p->input_eof = false;
return;
}
if (p->avrctx && !(!p->is_resampling && new_rate == p->in_rate)) {
AVRational r = av_d2q(p->playback_speed * p->in_rate_user / p->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(p->avrctx, r.den - r.num, r.den) >= 0) {
new_rate = p->in_rate;
p->is_resampling = true;
}
}
bool need_reinit = fabs(new_rate / (double)p->in_rate - 1) > 0.01;
if (need_reinit && new_rate != p->in_rate) {
// Before reconfiguring, drain the audio that is still buffered
// in the resampler.
filter_resample(p, NULL);
// Reinitialize resampler.
configure_lavrr(p, false);
p->output_eof = false;
if (p->output)
return; // need to read output before continuing filtering
}
filter_resample(p, p->input);
TA_FREEP(&p->input);
p->input_eof = false;
}
// Queue input. If true, ownership of in passes to mp_aconverted and the input
// was accepted. Otherwise, return false and reject in.
// in==NULL means trigger EOF.
bool mp_aconverter_write_input(struct mp_aconverter *p, struct mp_aframe *in)
{
if (p->input || p->input_eof)
return false;
p->input = in;
p->input_eof = !in;
return true;
}
// Return output frame, or NULL if nothing available.
// *eof is set to true if NULL is returned, and it was due to EOF.
struct mp_aframe *mp_aconverter_read_output(struct mp_aconverter *p, bool *eof)
{
*eof = false;
filter(p);
if (p->output) {
struct mp_aframe *out = p->output;
p->output = NULL;
return out;
}
*eof = p->output_eof;
p->output_eof = false;
return NULL;
}
double mp_aconverter_get_latency(struct mp_aconverter *p)
{
double delay = get_delay(p);
if (p->input)
delay += mp_aframe_duration(p->input);
// In theory this is influenced by playback speed, but other parts of the
// player get it wrong anyway.
if (p->output)
delay += mp_aframe_duration(p->output);
return delay;
}
static void destroy_aconverter(void *ptr)
{
struct mp_aconverter *p = ptr;
close_lavrr(p);
talloc_free(p->input);
talloc_free(p->output);
}
// If opts is not NULL, the pointer must be valid for the lifetime of the
// mp_aconverter.
struct mp_aconverter *mp_aconverter_create(struct mpv_global *global,
struct mp_log *log,
const struct mp_resample_opts *opts)
{
struct mp_aconverter *p = talloc_zero(NULL, struct mp_aconverter);
p->log = log;
p->global = global;
static const struct mp_resample_opts defs = MP_RESAMPLE_OPTS_DEF;
p->opts = opts ? opts : &defs;
p->reorder_buffer = mp_aframe_pool_create(p);
p->out_pool = mp_aframe_pool_create(p);
talloc_set_destructor(p, destroy_aconverter);
return p;
}
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