mpv/audio/aframe.c

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audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
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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/frame.h>
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#include <libavutil/mem.h>
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
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#include "common/common.h"
#include "chmap.h"
#include "fmt-conversion.h"
#include "format.h"
#include "aframe.h"
struct mp_aframe {
AVFrame *av_frame;
// We support channel layouts different from AVFrame channel masks
struct mp_chmap chmap;
// We support spdif formats, which are allocated as AV_SAMPLE_FMT_S16.
int format;
double pts;
};
static void free_frame(void *ptr)
{
struct mp_aframe *frame = ptr;
av_frame_free(&frame->av_frame);
}
struct mp_aframe *mp_aframe_create(void)
{
struct mp_aframe *frame = talloc_zero(NULL, struct mp_aframe);
frame->pts = MP_NOPTS_VALUE;
frame->av_frame = av_frame_alloc();
if (!frame->av_frame)
abort();
talloc_set_destructor(frame, free_frame);
return frame;
}
struct mp_aframe *mp_aframe_new_ref(struct mp_aframe *frame)
{
if (!frame)
return NULL;
struct mp_aframe *dst = mp_aframe_create();
dst->chmap = frame->chmap;
dst->format = frame->format;
dst->pts = frame->pts;
if (mp_aframe_is_allocated(frame)) {
if (av_frame_ref(dst->av_frame, frame->av_frame) < 0)
abort();
} else {
// av_frame_ref() would fail.
mp_aframe_config_copy(dst, frame);
}
return dst;
}
// Revert to state after mp_aframe_create().
void mp_aframe_reset(struct mp_aframe *frame)
{
av_frame_unref(frame->av_frame);
frame->chmap.num = 0;
frame->format = 0;
frame->pts = MP_NOPTS_VALUE;
}
// Remove all actual audio data and leave only the metadata.
void mp_aframe_unref_data(struct mp_aframe *frame)
{
// In a fucked up way, this is less complex than just unreffing the data.
struct mp_aframe *tmp = mp_aframe_create();
MPSWAP(struct mp_aframe, *tmp, *frame);
mp_aframe_reset(frame);
mp_aframe_config_copy(frame, tmp);
talloc_free(tmp);
}
// Return a new reference to the data in av_frame. av_frame itself is not
// touched. Returns NULL if not representable, or if input is NULL.
// Does not copy the timestamps.
struct mp_aframe *mp_aframe_from_avframe(struct AVFrame *av_frame)
{
if (!av_frame || av_frame->width > 0 || av_frame->height > 0)
return NULL;
int format = af_from_avformat(av_frame->format);
if (!format && av_frame->format != AV_SAMPLE_FMT_NONE)
return NULL;
struct mp_aframe *frame = mp_aframe_create();
// This also takes care of forcing refcounting.
if (av_frame_ref(frame->av_frame, av_frame) < 0)
abort();
frame->format = format;
mp_chmap_from_lavc(&frame->chmap, frame->av_frame->channel_layout);
#if LIBAVUTIL_VERSION_MICRO >= 100
// FFmpeg being a stupid POS again
if (frame->chmap.num != frame->av_frame->channels)
mp_chmap_from_channels(&frame->chmap, av_frame->channels);
#endif
return frame;
}
// Return a new reference to the data in frame. Returns NULL is not
// representable (), or if input is NULL.
// Does not copy the timestamps.
struct AVFrame *mp_aframe_to_avframe(struct mp_aframe *frame)
{
if (!frame)
return NULL;
if (af_to_avformat(frame->format) != frame->av_frame->format)
return NULL;
if (!mp_chmap_is_lavc(&frame->chmap))
return NULL;
return av_frame_clone(frame->av_frame);
}
struct AVFrame *mp_aframe_to_avframe_and_unref(struct mp_aframe *frame)
{
AVFrame *av = mp_aframe_to_avframe(frame);
talloc_free(frame);
return av;
}
// You must not use this.
struct AVFrame *mp_aframe_get_raw_avframe(struct mp_aframe *frame)
{
return frame->av_frame;
}
// Return whether it has associated audio data. (If not, metadata only.)
bool mp_aframe_is_allocated(struct mp_aframe *frame)
{
return frame->av_frame->buf[0] || frame->av_frame->extended_data[0];
}
// Clear dst, and then copy the configuration to it.
void mp_aframe_config_copy(struct mp_aframe *dst, struct mp_aframe *src)
{
mp_aframe_reset(dst);
dst->chmap = src->chmap;
dst->format = src->format;
dst->pts = src->pts;
if (av_frame_copy_props(dst->av_frame, src->av_frame) < 0)
abort();
dst->av_frame->format = src->av_frame->format;
dst->av_frame->channel_layout = src->av_frame->channel_layout;
#if LIBAVUTIL_VERSION_MICRO >= 100
// FFmpeg being a stupid POS again
dst->av_frame->channels = src->av_frame->channels;
#endif
}
// Return whether a and b use the same physical audio format. Extra metadata
// such as PTS, per-frame signalling, and AVFrame side data is not compared.
bool mp_aframe_config_equals(struct mp_aframe *a, struct mp_aframe *b)
{
struct mp_chmap ca = {0}, cb = {0};
mp_aframe_get_chmap(a, &ca);
mp_aframe_get_chmap(b, &cb);
return mp_chmap_equals(&ca, &cb) &&
mp_aframe_get_rate(a) == mp_aframe_get_rate(b) &&
mp_aframe_get_format(a) == mp_aframe_get_format(b);
}
// Return whether all required format fields have been set.
bool mp_aframe_config_is_valid(struct mp_aframe *frame)
{
return frame->format && frame->chmap.num && frame->av_frame->sample_rate;
}
// Return the pointer to the first sample for each plane. The pointers stay
// valid until the next call that mutates frame somehow. You must not write to
// the audio data. Returns NULL if no frame allocated.
uint8_t **mp_aframe_get_data_ro(struct mp_aframe *frame)
{
return mp_aframe_is_allocated(frame) ? frame->av_frame->extended_data : NULL;
}
// Like mp_aframe_get_data_ro(), but you can write to the audio data.
// Additionally, it will return NULL if copy-on-write fails.
uint8_t **mp_aframe_get_data_rw(struct mp_aframe *frame)
{
if (!mp_aframe_is_allocated(frame))
return NULL;
if (av_frame_make_writable(frame->av_frame) < 0)
return NULL;
return frame->av_frame->extended_data;
}
int mp_aframe_get_format(struct mp_aframe *frame)
{
return frame->format;
}
bool mp_aframe_get_chmap(struct mp_aframe *frame, struct mp_chmap *out)
{
if (!mp_chmap_is_valid(&frame->chmap))
return false;
*out = frame->chmap;
return true;
}
int mp_aframe_get_channels(struct mp_aframe *frame)
{
return frame->chmap.num;
}
int mp_aframe_get_rate(struct mp_aframe *frame)
{
return frame->av_frame->sample_rate;
}
int mp_aframe_get_size(struct mp_aframe *frame)
{
return frame->av_frame->nb_samples;
}
double mp_aframe_get_pts(struct mp_aframe *frame)
{
return frame->pts;
}
bool mp_aframe_set_format(struct mp_aframe *frame, int format)
{
if (mp_aframe_is_allocated(frame))
return false;
enum AVSampleFormat av_format = af_to_avformat(format);
if (av_format == AV_SAMPLE_FMT_NONE && format) {
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
if (!af_fmt_is_spdif(format))
return false;
av_format = AV_SAMPLE_FMT_S16;
}
frame->format = format;
frame->av_frame->format = av_format;
return true;
}
bool mp_aframe_set_chmap(struct mp_aframe *frame, struct mp_chmap *in)
{
if (!mp_chmap_is_valid(in) && !mp_chmap_is_empty(in))
return false;
if (mp_aframe_is_allocated(frame) && in->num != frame->chmap.num)
return false;
uint64_t lavc_layout = mp_chmap_to_lavc_unchecked(in);
if (!lavc_layout && in->num)
return false;
frame->chmap = *in;
frame->av_frame->channel_layout = lavc_layout;
#if LIBAVUTIL_VERSION_MICRO >= 100
// FFmpeg being a stupid POS again
frame->av_frame->channels = frame->chmap.num;
#endif
return true;
}
bool mp_aframe_set_rate(struct mp_aframe *frame, int rate)
{
if (rate < 1 && rate > 10000000)
return false;
frame->av_frame->sample_rate = rate;
return true;
}
bool mp_aframe_set_size(struct mp_aframe *frame, int samples)
{
if (!mp_aframe_is_allocated(frame) || mp_aframe_get_size(frame) < samples)
return false;
frame->av_frame->nb_samples = MPMAX(samples, 0);
return true;
}
void mp_aframe_set_pts(struct mp_aframe *frame, double pts)
{
frame->pts = pts;
}
// Return number of data pointers.
int mp_aframe_get_planes(struct mp_aframe *frame)
{
return af_fmt_is_planar(mp_aframe_get_format(frame))
? mp_aframe_get_channels(frame) : 1;
}
// Return number of bytes between 2 consecutive samples on the same plane.
size_t mp_aframe_get_sstride(struct mp_aframe *frame)
{
int format = mp_aframe_get_format(frame);
return af_fmt_to_bytes(format) *
(af_fmt_is_planar(format) ? 1 : mp_aframe_get_channels(frame));
}
// Set data to the audio after the given number of samples (i.e. slice it).
void mp_aframe_skip_samples(struct mp_aframe *f, int samples)
{
assert(samples >= 0 && samples <= mp_aframe_get_size(f));
int num_planes = mp_aframe_get_planes(f);
size_t sstride = mp_aframe_get_sstride(f);
for (int n = 0; n < num_planes; n++)
f->av_frame->extended_data[n] += samples * sstride;
f->av_frame->nb_samples -= samples;
if (f->pts != MP_NOPTS_VALUE)
f->pts += samples / (double)mp_aframe_get_rate(f);
}
// Return the timestamp of the sample just after the end of this frame.
double mp_aframe_end_pts(struct mp_aframe *f)
{
int rate = mp_aframe_get_rate(f);
if (f->pts == MP_NOPTS_VALUE || rate < 1)
return MP_NOPTS_VALUE;
return f->pts + f->av_frame->nb_samples / (double)rate;
}
// Return the duration in seconds of the frame (0 if invalid).
double mp_aframe_duration(struct mp_aframe *f)
{
int rate = mp_aframe_get_rate(f);
if (rate < 1)
return 0;
return f->av_frame->nb_samples / (double)rate;
}
// Clip the given frame to the given timestamp range. Adjusts the frame size
// and timestamp.
void mp_aframe_clip_timestamps(struct mp_aframe *f, double start, double end)
{
double f_end = mp_aframe_end_pts(f);
int rate = mp_aframe_get_rate(f);
if (f_end == MP_NOPTS_VALUE)
return;
if (end != MP_NOPTS_VALUE) {
if (f_end >= end) {
if (f->pts >= end) {
f->av_frame->nb_samples = 0;
} else {
int new = (end - f->pts) * rate;
f->av_frame->nb_samples = MPCLAMP(new, 0, f->av_frame->nb_samples);
}
}
}
if (start != MP_NOPTS_VALUE) {
if (f->pts < start) {
if (f_end <= start) {
f->av_frame->nb_samples = 0;
f->pts = f_end;
} else {
int skip = (start - f->pts) * rate;
skip = MPCLAMP(skip, 0, f->av_frame->nb_samples);
mp_aframe_skip_samples(f, skip);
}
}
}
}
struct mp_aframe_pool {
AVBufferPool *avpool;
int element_size;
};
struct mp_aframe_pool *mp_aframe_pool_create(void *ta_parent)
{
return talloc_zero(ta_parent, struct mp_aframe_pool);
}
static void mp_aframe_pool_destructor(void *p)
{
struct mp_aframe_pool *pool = p;
av_buffer_pool_uninit(&pool->avpool);
}
// Like mp_aframe_allocate(), but use the pool to allocate data.
int mp_aframe_pool_allocate(struct mp_aframe_pool *pool, struct mp_aframe *frame,
int samples)
{
int planes = mp_aframe_get_planes(frame);
size_t sstride = mp_aframe_get_sstride(frame);
int plane_size = MP_ALIGN_UP(sstride * MPMAX(samples, 1), 32);
int size = plane_size * planes;
if (size <= 0 || mp_aframe_is_allocated(frame))
return -1;
if (!pool->avpool || size > pool->element_size) {
size_t alloc = ta_calc_prealloc_elems(size);
if (alloc >= INT_MAX)
return -1;
av_buffer_pool_uninit(&pool->avpool);
pool->element_size = alloc;
pool->avpool = av_buffer_pool_init(pool->element_size, NULL);
if (!pool->avpool)
return -1;
talloc_set_destructor(pool, mp_aframe_pool_destructor);
}
// Yes, you have to do all this shit manually.
// At least it's less stupid than av_frame_get_buffer(), which just wipes
// the entire frame struct on error for no reason.
AVFrame *av_frame = frame->av_frame;
if (av_frame->extended_data != av_frame->data)
av_freep(&av_frame->extended_data); // sigh
av_frame->extended_data =
av_mallocz_array(planes, sizeof(av_frame->extended_data[0]));
if (!av_frame->extended_data)
abort();
av_frame->buf[0] = av_buffer_pool_get(pool->avpool);
if (!av_frame->buf[0])
return -1;
av_frame->linesize[0] = samples * sstride;
for (int n = 0; n < planes; n++)
av_frame->extended_data[n] = av_frame->buf[0]->data + n * plane_size;
av_frame->nb_samples = samples;
return 0;
}