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mpv/audio/audio.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 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/>.
*/
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#include <stdint.h>
#include <limits.h>
#include <stdlib.h>
#include <assert.h>
#include <libavutil/buffer.h>
#include <libavutil/frame.h>
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#include <libavutil/mem.h>
#include <libavutil/version.h>
#include "mpv_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;
if (data->pts != MP_NOPTS_VALUE)
data->pts += samples / (double)data->rate;
}
// Clip the given frame to the given timestamp range. Adjusts the frame size
// and timestamp.
void mp_audio_clip_timestamps(struct mp_audio *f, double start, double end)
{
if (f->pts == MP_NOPTS_VALUE || f->rate < 1)
return;
double f_end = f->pts + f->samples / (double)f->rate;
if (end != MP_NOPTS_VALUE) {
if (f_end >= end) {
if (f->pts >= end) {
f->samples = 0;
} else {
int new = (end - f->pts) * f->rate;
f->samples = MPCLAMP(new, 0, f->samples);
}
}
}
if (start != MP_NOPTS_VALUE) {
if (f->pts < start) {
if (f_end <= start) {
f->samples = 0;
f->pts = f_end;
} else {
int skip = (start - f->pts) * f->rate;
skip = MPCLAMP(skip, 0, f->samples);
mp_audio_skip_samples(f, skip);
}
}
}
}
// 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)
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{
bool ok = true;
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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)) {
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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);
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}
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))
goto fail;
for (int n = 0; n < AV_NUM_DATA_POINTERS + avframe->nb_extended_buf; n++) {
AVBufferRef *buf = n < AV_NUM_DATA_POINTERS ? avframe->buf[n]
: avframe->extended_buf[n - AV_NUM_DATA_POINTERS];
if (!buf)
break;
if (n >= MP_NUM_CHANNELS)
goto fail;
new->allocated[n] = av_buffer_ref(buf);
if (!new->allocated[n])
goto fail;
}
for (int n = 0; n < new->num_planes; n++)
new->planes[n] = avframe->extended_data[n];
new->samples = avframe->nb_samples;
return new;
fail:
talloc_free(new);
av_frame_free(&tmp);
return NULL;
}
// Returns NULL on failure. The input is always unreffed.
struct AVFrame *mp_audio_to_avframe_and_unref(struct mp_audio *frame)
{
struct AVFrame *avframe = av_frame_alloc();
if (!avframe)
goto fail;
avframe->nb_samples = frame->samples;
avframe->format = af_to_avformat(frame->format);
if (avframe->format == AV_SAMPLE_FMT_NONE)
goto fail;
avframe->channel_layout = mp_chmap_to_lavc(&frame->channels);
if (!avframe->channel_layout)
goto fail;
#if LIBAVUTIL_VERSION_MICRO >= 100
// FFmpeg being a stupid POS (but I respect it)
avframe->channels = frame->channels.num;
#endif
avframe->sample_rate = frame->rate;
if (frame->num_planes > AV_NUM_DATA_POINTERS) {
avframe->extended_data =
av_mallocz_array(frame->num_planes, sizeof(avframe->extended_data[0]));
int extbufs = frame->num_planes - AV_NUM_DATA_POINTERS;
avframe->extended_buf =
av_mallocz_array(extbufs, sizeof(avframe->extended_buf[0]));
if (!avframe->extended_data || !avframe->extended_buf)
goto fail;
avframe->nb_extended_buf = extbufs;
}
for (int p = 0; p < frame->num_planes; p++)
avframe->extended_data[p] = frame->planes[p];
avframe->linesize[0] = frame->samples * frame->sstride;
for (int p = 0; p < AV_NUM_DATA_POINTERS; p++)
avframe->data[p] = avframe->extended_data[p];
for (int p = 0; p < frame->num_planes; p++) {
if (!frame->allocated[p])
break;
AVBufferRef *nref = av_buffer_ref(frame->allocated[p]);
if (!nref)
goto fail;
if (p < AV_NUM_DATA_POINTERS) {
avframe->buf[p] = nref;
} else {
avframe->extended_buf[p - AV_NUM_DATA_POINTERS] = nref;
}
}
// Force refcounted frame.
if (!avframe->buf[0]) {
AVFrame *tmp = av_frame_alloc();
if (!tmp)
goto fail;
if (av_frame_ref(tmp, avframe) < 0)
goto fail;
av_frame_free(&avframe);
avframe = tmp;
}
talloc_free(frame);
return avframe;
fail:
av_frame_free(&avframe);
talloc_free(frame);
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;
}