1
0
mirror of https://github.com/mpv-player/mpv synced 2024-12-25 08:12:17 +00:00
mpv/audio/audio.c
wm4 091bfa3abf audio/filter: remove some useless filters
All of these filters are considered not useful anymore by us. Some have
replacements in libavfilter (useable through af_lavfi).

af_center, af_extrastereo, af_karaoke, af_sinesuppress, af_sub,
af_surround, af_sweep: pretty simple and useless filters which probably
nobody ever wants.

af_ladspa: has a replacement in libavfilter.

af_hrtf: the algorithm doesn't work properly on most sources, and the
implementation was buggy and complicated. (The filter was inherited from
MPlayer; but even in mpv times we had to apply fixes that fixed major
issues with added noise.) There is a ladspa filter if you still want to
use it.

af_export: I'm not even sure what this is supposed to do. Possibly it
was meant for GUIs rendering audio visualizations, but it couldn't
really work well. For example, the size of the audio depended on the
samplerate (fixed number of samples only), and it couldn't retrieve the
complete audio, only fragments. If this is really needed for GUIs, mpv
should add native visualization, or a proper API for it.
2015-09-03 23:55:36 +02:00

429 lines
13 KiB
C

/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdint.h>
#include <limits.h>
#include <stdlib.h>
#include <assert.h>
#include <libavutil/buffer.h>
#include <libavutil/frame.h>
#include <libavutil/version.h>
#include "talloc.h"
#include "common/common.h"
#include "fmt-conversion.h"
#include "audio.h"
static void update_redundant_info(struct mp_audio *mpa)
{
assert(mp_chmap_is_empty(&mpa->channels) ||
mp_chmap_is_valid(&mpa->channels));
mpa->nch = mpa->channels.num;
mpa->bps = af_fmt_to_bytes(mpa->format);
if (af_fmt_is_planar(mpa->format)) {
mpa->spf = 1;
mpa->num_planes = mpa->nch;
mpa->sstride = mpa->bps;
} else {
mpa->spf = mpa->nch;
mpa->num_planes = 1;
mpa->sstride = mpa->bps * mpa->nch;
}
}
void mp_audio_set_format(struct mp_audio *mpa, int format)
{
mpa->format = format;
update_redundant_info(mpa);
}
void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels)
{
mp_chmap_from_channels(&mpa->channels, num_channels);
update_redundant_info(mpa);
}
void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap)
{
mpa->channels = *chmap;
update_redundant_info(mpa);
}
void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src)
{
dst->format = src->format;
dst->channels = src->channels;
dst->rate = src->rate;
update_redundant_info(dst);
}
bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b)
{
return a->format == b->format && a->rate == b->rate &&
mp_chmap_equals(&a->channels, &b->channels);
}
bool mp_audio_config_valid(const struct mp_audio *mpa)
{
return mp_chmap_is_valid(&mpa->channels) && af_fmt_is_valid(mpa->format)
&& mpa->rate >= 1 && mpa->rate < 10000000;
}
char *mp_audio_config_to_str_buf(char *buf, size_t buf_sz, struct mp_audio *mpa)
{
char ch[128];
mp_chmap_to_str_buf(ch, sizeof(ch), &mpa->channels);
char *hr_ch = mp_chmap_to_str_hr(&mpa->channels);
if (strcmp(hr_ch, ch) != 0)
mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch);
snprintf(buf, buf_sz, "%dHz %s %dch %s", mpa->rate,
ch, mpa->channels.num, af_fmt_to_str(mpa->format));
return buf;
}
void mp_audio_force_interleaved_format(struct mp_audio *mpa)
{
if (af_fmt_is_planar(mpa->format))
mp_audio_set_format(mpa, af_fmt_from_planar(mpa->format));
}
// Return used size of a plane. (The size is the same for all planes.)
int mp_audio_psize(struct mp_audio *mpa)
{
return mpa->samples * mpa->sstride;
}
void mp_audio_set_null_data(struct mp_audio *mpa)
{
for (int n = 0; n < MP_NUM_CHANNELS; n++) {
mpa->planes[n] = NULL;
mpa->allocated[n] = NULL;
}
mpa->samples = 0;
}
static int get_plane_size(const struct mp_audio *mpa, int samples)
{
if (samples < 0 || !mp_audio_config_valid(mpa))
return -1;
if (samples >= INT_MAX / mpa->sstride)
return -1;
return MPMAX(samples * mpa->sstride, 1);
}
static void mp_audio_destructor(void *ptr)
{
struct mp_audio *mpa = ptr;
for (int n = 0; n < MP_NUM_CHANNELS; n++)
av_buffer_unref(&mpa->allocated[n]);
}
/* Reallocate the data stored in mpa->planes[n] so that enough samples are
* available on every plane. The previous data is kept (for the smallest
* common number of samples before/after resize).
*
* mpa->samples is not set or used.
*
* This function is flexible enough to handle format and channel layout
* changes. In these cases, all planes are reallocated as needed. Unused
* planes are freed.
*
* mp_audio_realloc(mpa, 0) will still yield non-NULL for mpa->data[n].
*
* Allocated data is implicitly freed on talloc_free(mpa).
*/
void mp_audio_realloc(struct mp_audio *mpa, int samples)
{
int size = get_plane_size(mpa, samples);
if (size < 0)
abort(); // oom or invalid parameters
for (int n = 0; n < mpa->num_planes; n++) {
if (!mpa->allocated[n] || size != mpa->allocated[n]->size) {
if (av_buffer_realloc(&mpa->allocated[n], size) < 0)
abort(); // OOM
}
mpa->planes[n] = mpa->allocated[n]->data;
}
for (int n = mpa->num_planes; n < MP_NUM_CHANNELS; n++) {
av_buffer_unref(&mpa->allocated[n]);
mpa->planes[n] = NULL;
}
talloc_set_destructor(mpa, mp_audio_destructor);
}
// Like mp_audio_realloc(), but only reallocate if the audio grows in size.
// If the buffer is reallocated, also preallocate.
void mp_audio_realloc_min(struct mp_audio *mpa, int samples)
{
if (samples > mp_audio_get_allocated_size(mpa)) {
size_t alloc = ta_calc_prealloc_elems(samples);
if (alloc > INT_MAX)
abort(); // oom
mp_audio_realloc(mpa, alloc);
}
}
/* Get the size allocated for the data, in number of samples. If the allocated
* size isn't on sample boundaries (e.g. after format changes), the returned
* sample number is a rounded down value.
*
* Note that this only works in situations where mp_audio_realloc() also works!
*/
int mp_audio_get_allocated_size(struct mp_audio *mpa)
{
int size = 0;
for (int n = 0; n < mpa->num_planes; n++) {
for (int i = 0; i < MP_NUM_CHANNELS && mpa->allocated[i]; i++) {
uint8_t *start = mpa->allocated[i]->data;
uint8_t *end = start + mpa->allocated[i]->size;
uint8_t *plane = mpa->planes[n];
if (plane >= start && plane < end) {
int s = MPMIN((end - plane) / mpa->sstride, INT_MAX);
size = n == 0 ? s : MPMIN(size, s);
goto next;
}
}
return 0; // plane is not covered by any buffer
next: ;
}
return size;
}
// Clear the samples [start, start + length) with silence.
void mp_audio_fill_silence(struct mp_audio *mpa, int start, int length)
{
assert(start >= 0 && length >= 0 && start + length <= mpa->samples);
int offset = start * mpa->sstride;
int size = length * mpa->sstride;
for (int n = 0; n < mpa->num_planes; n++) {
if (n > 0 && mpa->planes[n] == mpa->planes[0])
continue; // silly optimization for special cases
af_fill_silence((char *)mpa->planes[n] + offset, size, mpa->format);
}
}
// All integer parameters are in samples.
// dst and src can overlap.
void mp_audio_copy(struct mp_audio *dst, int dst_offset,
struct mp_audio *src, int src_offset, int length)
{
assert(mp_audio_config_equals(dst, src));
assert(length >= 0);
assert(dst_offset >= 0 && dst_offset + length <= dst->samples);
assert(src_offset >= 0 && src_offset + length <= src->samples);
for (int n = 0; n < dst->num_planes; n++) {
memmove((char *)dst->planes[n] + dst_offset * dst->sstride,
(char *)src->planes[n] + src_offset * src->sstride,
length * dst->sstride);
}
}
// Copy fields that describe characteristics of the audio frame, but which are
// not part of the core format (format/channels/rate), and not part of the
// data (samples).
void mp_audio_copy_attributes(struct mp_audio *dst, struct mp_audio *src)
{
// nothing yet
}
// Set data to the audio after the given number of samples (i.e. slice it).
void mp_audio_skip_samples(struct mp_audio *data, int samples)
{
assert(samples >= 0 && samples <= data->samples);
for (int n = 0; n < data->num_planes; n++)
data->planes[n] = (uint8_t *)data->planes[n] + samples * data->sstride;
data->samples -= samples;
}
// Return false if the frame data is shared, true otherwise.
// Will return true for non-refcounted frames.
bool mp_audio_is_writeable(struct mp_audio *data)
{
bool ok = true;
for (int n = 0; n < MP_NUM_CHANNELS; n++) {
if (data->allocated[n])
ok &= av_buffer_is_writable(data->allocated[n]);
}
return ok;
}
static void mp_audio_steal_data(struct mp_audio *dst, struct mp_audio *src)
{
talloc_set_destructor(dst, mp_audio_destructor);
mp_audio_destructor(dst);
*dst = *src;
talloc_set_destructor(src, NULL);
talloc_free(src);
}
// Make sure the frame owns the audio data, and if not, copy the data.
// Return negative value on failure (which means it can't be made writeable).
// Non-refcounted frames are always considered writeable.
int mp_audio_make_writeable(struct mp_audio *data)
{
if (!mp_audio_is_writeable(data)) {
struct mp_audio *new = talloc(NULL, struct mp_audio);
*new = *data;
mp_audio_set_null_data(new); // use format only
mp_audio_realloc(new, data->samples);
new->samples = data->samples;
mp_audio_copy(new, 0, data, 0, data->samples);
mp_audio_steal_data(data, new);
}
return 0;
}
struct mp_audio *mp_audio_from_avframe(struct AVFrame *avframe)
{
AVFrame *tmp = NULL;
struct mp_audio *new = talloc_zero(NULL, struct mp_audio);
talloc_set_destructor(new, mp_audio_destructor);
mp_audio_set_format(new, af_from_avformat(avframe->format));
struct mp_chmap lavc_chmap;
mp_chmap_from_lavc(&lavc_chmap, avframe->channel_layout);
#if LIBAVUTIL_VERSION_MICRO >= 100
// FFmpeg being special again
if (lavc_chmap.num != avframe->channels)
mp_chmap_from_channels(&lavc_chmap, avframe->channels);
#endif
new->rate = avframe->sample_rate;
mp_audio_set_channels(new, &lavc_chmap);
// Force refcounted frame.
if (!avframe->buf[0]) {
tmp = av_frame_alloc();
if (!tmp)
goto fail;
if (av_frame_ref(tmp, avframe) < 0)
goto fail;
avframe = tmp;
}
// If we can't handle the format (e.g. too many channels), bail out.
if (!mp_audio_config_valid(new) || avframe->nb_extended_buf)
goto fail;
for (int n = 0; n < AV_NUM_DATA_POINTERS; n++) {
if (!avframe->buf[n])
break;
if (n >= MP_NUM_CHANNELS)
goto fail;
new->allocated[n] = av_buffer_ref(avframe->buf[n]);
if (!new->allocated[n])
goto fail;
}
for (int n = 0; n < new->num_planes; n++)
new->planes[n] = avframe->data[n];
new->samples = avframe->nb_samples;
return new;
fail:
talloc_free(new);
av_frame_free(&tmp);
return NULL;
}
struct mp_audio_pool {
AVBufferPool *avpool;
int element_size;
};
struct mp_audio_pool *mp_audio_pool_create(void *ta_parent)
{
return talloc_zero(ta_parent, struct mp_audio_pool);
}
static void mp_audio_pool_destructor(void *p)
{
struct mp_audio_pool *pool = p;
av_buffer_pool_uninit(&pool->avpool);
}
// Allocate data using the given format and number of samples.
// Returns NULL on error.
struct mp_audio *mp_audio_pool_get(struct mp_audio_pool *pool,
const struct mp_audio *fmt, int samples)
{
int size = get_plane_size(fmt, samples);
if (size < 0)
return NULL;
if (!pool->avpool || size > pool->element_size) {
size_t alloc = ta_calc_prealloc_elems(size);
if (alloc >= INT_MAX)
return NULL;
av_buffer_pool_uninit(&pool->avpool);
pool->element_size = alloc;
pool->avpool = av_buffer_pool_init(pool->element_size, NULL);
if (!pool->avpool)
return NULL;
talloc_set_destructor(pool, mp_audio_pool_destructor);
}
struct mp_audio *new = talloc_ptrtype(NULL, new);
talloc_set_destructor(new, mp_audio_destructor);
*new = *fmt;
mp_audio_set_null_data(new);
new->samples = samples;
for (int n = 0; n < new->num_planes; n++) {
new->allocated[n] = av_buffer_pool_get(pool->avpool);
if (!new->allocated[n]) {
talloc_free(new);
return NULL;
}
new->planes[n] = new->allocated[n]->data;
}
return new;
}
// Return a copy of the given frame.
// Returns NULL on error.
struct mp_audio *mp_audio_pool_new_copy(struct mp_audio_pool *pool,
struct mp_audio *frame)
{
struct mp_audio *new = mp_audio_pool_get(pool, frame, frame->samples);
if (new) {
mp_audio_copy(new, 0, frame, 0, new->samples);
mp_audio_copy_attributes(new, frame);
}
return new;
}
// Exactly like mp_audio_make_writeable(), but get the data from the pool.
int mp_audio_pool_make_writeable(struct mp_audio_pool *pool,
struct mp_audio *data)
{
if (mp_audio_is_writeable(data))
return 0;
struct mp_audio *new = mp_audio_pool_get(pool, data, data->samples);
if (!new)
return -1;
mp_audio_copy(new, 0, data, 0, data->samples);
mp_audio_copy_attributes(new, data);
mp_audio_steal_data(data, new);
return 0;
}