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;
double speed;
};
struct avframe_opaque {
double speed;
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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};
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->av_frame = av_frame_alloc();
if (!frame->av_frame)
abort();
talloc_set_destructor(frame, free_frame);
mp_aframe_reset(frame);
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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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;
dst->speed = frame->speed;
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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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;
frame->speed = 1.0;
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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}
// 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
if (av_frame->opaque_ref) {
struct avframe_opaque *op = (void *)av_frame->opaque_ref->data;
frame->speed = op->speed;
}
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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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;
if (!frame->av_frame->opaque_ref && frame->speed != 1.0) {
frame->av_frame->opaque_ref =
av_buffer_alloc(sizeof(struct avframe_opaque));
if (!frame->av_frame->opaque_ref)
return NULL;
struct avframe_opaque *op = (void *)frame->av_frame->opaque_ref->data;
op->speed = frame->speed;
}
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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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;
mp_aframe_copy_attributes(dst, src);
dst->av_frame->sample_rate = src->av_frame->sample_rate;
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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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
}
// Copy "soft" attributes from src to dst, excluding things which affect
// frame allocation and organization.
void mp_aframe_copy_attributes(struct mp_aframe *dst, struct mp_aframe *src)
{
dst->pts = src->pts;
dst->speed = src->speed;
int rate = dst->av_frame->sample_rate;
if (av_frame_copy_props(dst->av_frame, src->av_frame) < 0)
abort();
dst->av_frame->sample_rate = rate;
}
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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// 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.
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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)
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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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;
}
// Set a speed factor. This is multiplied with the sample rate to get the
// "effective" samplerate (mp_aframe_get_effective_rate()), which will be used
// to do PTS calculations. If speed!=1.0, the PTS values always refer to the
// original PTS (before changing speed), and if you want reasonably continuous
// PTS between frames, you need to use the effective samplerate.
void mp_aframe_set_speed(struct mp_aframe *frame, double factor)
{
frame->speed = factor;
}
// Adjust current speed factor.
void mp_aframe_mul_speed(struct mp_aframe *frame, double factor)
{
frame->speed *= factor;
}
double mp_aframe_get_speed(struct mp_aframe *frame)
{
return frame->speed;
}
// Matters for speed changed frames (such as a frame which has been resampled
// to play at a different speed).
// Return the sample rate at which the frame would have to be played to result
// in the same duration as the original frame before the speed change.
// This is used for A/V sync.
double mp_aframe_get_effective_rate(struct mp_aframe *frame)
{
return mp_aframe_get_rate(frame) / frame->speed;
}
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
// 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));
}
// Return total number of samples on each plane.
int mp_aframe_get_total_plane_samples(struct mp_aframe *frame)
{
return frame->av_frame->nb_samples *
(af_fmt_is_planar(mp_aframe_get_format(frame))
? 1 : mp_aframe_get_channels(frame));
}
char *mp_aframe_format_str_buf(char *buf, size_t buf_size, struct mp_aframe *fmt)
{
char ch[128];
mp_chmap_to_str_buf(ch, sizeof(ch), &fmt->chmap);
char *hr_ch = mp_chmap_to_str_hr(&fmt->chmap);
if (strcmp(hr_ch, ch) != 0)
mp_snprintf_cat(ch, sizeof(ch), " (%s)", hr_ch);
snprintf(buf, buf_size, "%dHz %s %dch %s", fmt->av_frame->sample_rate,
ch, fmt->chmap.num, af_fmt_to_str(fmt->format));
return buf;
}
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
// 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));
audio: work around ffmpeg being a piece of shit The "amultiply" filter crashes in AVX mode on unaligned access if an audio pointer is unaligned (on 32 or 64 bytes I assume). A requirement that audio data needs to be aligned isn't documented anywhere. In our case, the data is still sample- and channel-aligned, which is completely sane. Sure, you can imagine optimizations which make some algorithms even faster by requiring higher alignment. But, and this is a big but, you shouldn't crash api users because you just invented a new undocumented requirement. And even more importantly, your user-crashing optimization won't matter because it's just a trivial algorithm working on audio. You don't need to pretend to be an optimization devil, and nobody will give you a prize for this. But no, lets random make API users crash (and then probably blame them for it!) for something that wouldn't matter at all. Not to mention that they do "document" some requirements on _video_ data, yet their vf_crop probably can still produce unaligned video pointers. Oh how hilarious that the same documentation also talks about libswscale alignment requirements. (This is weird because libswscale is just one of many, many things which consume video data. Also did you know that zimg, written in C++ and using intrinsics, i.e. the antithesis to FFmpeg development, is much faster than libswscale, easier to use, and produces more correct results, even if you ignore that libswscale flat out doesn't support some very important features?) Fucking tired of this bullshit. Can't wait until someone comes up with a better framework than this... (well let's not write this out). Fix this by copying instead of adjusting the start pointer when skipping samples. This makes general operations slower just to fix interoperating with a single filter. Thank you for your "optimization", FFmpeg. Go die in a fire. Didn't check whether this is correct. It probably is? If the frame needs to be copied (due to COW), and memory allocation fails, it just silently (or audibly lol) doesn't skip samples, because a never-fail function can suddenly fail. Well, who cares. Fixes: #7141
2019-11-10 14:02:25 +00:00
if (av_frame_make_writable(f->av_frame) < 0)
return; // go complain to ffmpeg
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
int num_planes = mp_aframe_get_planes(f);
size_t sstride = mp_aframe_get_sstride(f);
audio: work around ffmpeg being a piece of shit The "amultiply" filter crashes in AVX mode on unaligned access if an audio pointer is unaligned (on 32 or 64 bytes I assume). A requirement that audio data needs to be aligned isn't documented anywhere. In our case, the data is still sample- and channel-aligned, which is completely sane. Sure, you can imagine optimizations which make some algorithms even faster by requiring higher alignment. But, and this is a big but, you shouldn't crash api users because you just invented a new undocumented requirement. And even more importantly, your user-crashing optimization won't matter because it's just a trivial algorithm working on audio. You don't need to pretend to be an optimization devil, and nobody will give you a prize for this. But no, lets random make API users crash (and then probably blame them for it!) for something that wouldn't matter at all. Not to mention that they do "document" some requirements on _video_ data, yet their vf_crop probably can still produce unaligned video pointers. Oh how hilarious that the same documentation also talks about libswscale alignment requirements. (This is weird because libswscale is just one of many, many things which consume video data. Also did you know that zimg, written in C++ and using intrinsics, i.e. the antithesis to FFmpeg development, is much faster than libswscale, easier to use, and produces more correct results, even if you ignore that libswscale flat out doesn't support some very important features?) Fucking tired of this bullshit. Can't wait until someone comes up with a better framework than this... (well let's not write this out). Fix this by copying instead of adjusting the start pointer when skipping samples. This makes general operations slower just to fix interoperating with a single filter. Thank you for your "optimization", FFmpeg. Go die in a fire. Didn't check whether this is correct. It probably is? If the frame needs to be copied (due to COW), and memory allocation fails, it just silently (or audibly lol) doesn't skip samples, because a never-fail function can suddenly fail. Well, who cares. Fixes: #7141
2019-11-10 14:02:25 +00:00
for (int n = 0; n < num_planes; n++) {
memmove(f->av_frame->extended_data[n],
f->av_frame->extended_data[n] + samples * sstride,
(f->av_frame->nb_samples - samples) * sstride);
}
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
f->av_frame->nb_samples -= samples;
if (f->pts != MP_NOPTS_VALUE)
f->pts += samples / mp_aframe_get_effective_rate(f);
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
}
// Return the timestamp of the sample just after the end of this frame.
double mp_aframe_end_pts(struct mp_aframe *f)
{
double rate = mp_aframe_get_effective_rate(f);
if (f->pts == MP_NOPTS_VALUE || rate <= 0)
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
return MP_NOPTS_VALUE;
return f->pts + f->av_frame->nb_samples / rate;
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
}
// Return the duration in seconds of the frame (0 if invalid).
double mp_aframe_duration(struct mp_aframe *f)
{
double rate = mp_aframe_get_effective_rate(f);
if (rate <= 0)
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
return 0;
return f->av_frame->nb_samples / rate;
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
}
// Clip the given frame to the given timestamp range. Adjusts the frame size
// and timestamp.
// Refuses to change spdif frames.
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
void mp_aframe_clip_timestamps(struct mp_aframe *f, double start, double end)
{
double f_end = mp_aframe_end_pts(f);
double rate = mp_aframe_get_effective_rate(f);
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 (f_end == MP_NOPTS_VALUE)
return;
if (af_fmt_is_spdif(mp_aframe_get_format(f)))
return;
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 (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);
}
}
}
}
bool mp_aframe_copy_samples(struct mp_aframe *dst, int dst_offset,
struct mp_aframe *src, int src_offset,
int samples)
{
if (!mp_aframe_config_equals(dst, src))
return false;
if (mp_aframe_get_size(dst) < dst_offset + samples ||
mp_aframe_get_size(src) < src_offset + samples)
return false;
uint8_t **s = mp_aframe_get_data_ro(src);
uint8_t **d = mp_aframe_get_data_rw(dst);
if (!s || !d)
return false;
int planes = mp_aframe_get_planes(dst);
size_t sstride = mp_aframe_get_sstride(dst);
for (int n = 0; n < planes; n++) {
memcpy(d[n] + dst_offset * sstride, s[n] + src_offset * sstride,
samples * sstride);
}
return true;
}
bool mp_aframe_set_silence(struct mp_aframe *f, int offset, int samples)
{
if (mp_aframe_get_size(f) < offset + samples)
return false;
int format = mp_aframe_get_format(f);
uint8_t **d = mp_aframe_get_data_rw(f);
if (!d)
return false;
int planes = mp_aframe_get_planes(f);
size_t sstride = mp_aframe_get_sstride(f);
for (int n = 0; n < planes; n++)
af_fill_silence(d[n] + offset * sstride, samples * sstride, format);
return true;
}
Implement backwards playback See manpage additions. This is a huge hack. You can bet there are shit tons of bugs. It's literally forcing square pegs into round holes. Hopefully, the manpage wall of text makes it clear enough that the whole shit can easily crash and burn. (Although it shouldn't literally crash. That would be a bug. It possibly _could_ start a fire by entering some sort of endless loop, not a literal one, just something where it tries to do work without making progress.) (Some obvious bugs I simply ignored for this initial version, but there's a number of potential bugs I can't even imagine. Normal playback should remain completely unaffected, though.) How this works is also described in the manpage. Basically, we demux in reverse, then we decode in reverse, then we render in reverse. The decoding part is the simplest: just reorder the decoder output. This weirdly integrates with the timeline/ordered chapter code, which also has special requirements on feeding the packets to the decoder in a non-straightforward way (it doesn't conflict, although a bugmessmass breaks correct slicing of segments, so EDL/ordered chapter playback is broken in backward direction). Backward demuxing is pretty involved. In theory, it could be much easier: simply iterating the usual demuxer output backward. But this just doesn't fit into our code, so there's a cthulhu nightmare of shit. To be specific, each stream (audio, video) is reversed separately. At least this means we can do backward playback within cached content (for example, you could play backwards in a live stream; on that note, it disables prefetching, which would lead to losing new live video, but this could be avoided). The fuckmess also meant that I didn't bother trying to support subtitles. Subtitles are a problem because they're "sparse" streams. They need to be "passively" demuxed: you don't try to read a subtitle packet, you demux audio and video, and then look whether there was a subtitle packet. This means to get subtitles for a time range, you need to know that you demuxed video and audio over this range, which becomes pretty messy when you demux audio and video backwards separately. Backward display is the most weird (and potentially buggy) part. To avoid that we need to touch a LOT of timing code, we negate all timestamps. The basic idea is that due to the navigation, all comparisons and subtractions of timestamps keep working, and you don't need to touch every single of them to "reverse" them. E.g.: bool before = pts_a < pts_b; would need to be: bool before = forward ? pts_a < pts_b : pts_a > pts_b; or: bool before = pts_a * dir < pts_b * dir; or if you, as it's implemented now, just do this after decoding: pts_a *= dir; pts_b *= dir; and then in the normal timing/renderer code: bool before = pts_a < pts_b; Consequently, we don't need many changes in the latter code. But some assumptions inhererently true for forward playback may have been broken anyway. What is mainly needed is fixing places where values are passed between positive and negative "domains". For example, seeking and timestamp user display always uses positive timestamps. The main mess is that it's not obvious which domain a given variable should or does use. Well, in my tests with a single file, it suddenly started to work when I did this. I'm honestly surprised that it did, and that I didn't have to change a single line in the timing code past decoder (just something minor to make external/cached text subtitles display). I committed it immediately while avoiding thinking about it. But there really likely are subtle problems of all sorts. As far as I'm aware, gstreamer also supports backward playback. When I looked at this years ago, I couldn't find a way to actually try this, and I didn't revisit it now. Back then I also read talk slides from the person who implemented it, and I'm not sure if and which ideas I might have taken from it. It's possible that the timestamp reversal is inspired by it, but I didn't check. (I think it claimed that it could avoid large changes by changing a sign?) VapourSynth has some sort of reverse function, which provides a backward view on a video. The function itself is trivial to implement, as VapourSynth aims to provide random access to video by frame numbers (so you just request decreasing frame numbers). From what I remember, it wasn't exactly fluid, but it worked. It's implemented by creating an index, and seeking to the target on demand, and a bunch of caching. mpv could use it, but it would either require using VapourSynth as demuxer and decoder for everything, or replacing the current file every time something is supposed to be played backwards. FFmpeg's libavfilter has reversal filters for audio and video. These require buffering the entire media data of the file, and don't really fit into mpv's architecture. It could be used by playing a libavfilter graph that also demuxes, but that's like VapourSynth but worse.
2019-05-18 00:10:51 +00:00
bool mp_aframe_reverse(struct mp_aframe *f)
{
int format = mp_aframe_get_format(f);
size_t bps = af_fmt_to_bytes(format);
if (!af_fmt_is_pcm(format) || bps > 16)
return false;
uint8_t **d = mp_aframe_get_data_rw(f);
if (!d)
return false;
int planes = mp_aframe_get_planes(f);
int samples = mp_aframe_get_size(f);
int channels = mp_aframe_get_channels(f);
size_t sstride = mp_aframe_get_sstride(f);
int plane_samples = channels;
if (af_fmt_is_planar(format))
plane_samples = 1;
for (int p = 0; p < planes; p++) {
for (int n = 0; n < samples / 2; n++) {
int s1_offset = n * sstride;
int s2_offset = (samples - 1 - n) * sstride;
for (int c = 0; c < plane_samples; c++) {
// Nobody said it'd be fast.
char tmp[16];
uint8_t *s1 = d[p] + s1_offset + c * bps;
uint8_t *s2 = d[p] + s2_offset + c * bps;
memcpy(tmp, s2, bps);
memcpy(s2, s1, bps);
memcpy(s1, tmp, bps);
}
}
}
return true;
}
int mp_aframe_approx_byte_size(struct mp_aframe *frame)
{
// God damn, AVFrame is too fucking annoying. Just go with the size that
// allocating a new frame would use.
int planes = mp_aframe_get_planes(frame);
size_t sstride = mp_aframe_get_sstride(frame);
int samples = frame->av_frame->nb_samples;
int plane_size = MP_ALIGN_UP(sstride * MPMAX(samples, 1), 32);
return plane_size * planes + sizeof(*frame);
}
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
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);
// FFmpeg hardcodes similar hidden possibly-requirements in a number of
// places: av_frame_get_buffer(), libavcodec's get_buffer(), mem.c,
// probably more.
int align_samples = MP_ALIGN_UP(MPMAX(samples, 1), 32);
int plane_size = MP_ALIGN_UP(sstride * align_samples, 64);
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
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;
}