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mpv/audio/decode/ad_lavc.c
Philip Langdale 4574dd5dc6 ffmpeg: update to handle deprecation of av_init_packet
This has been a long standing annoyance - ffmpeg is removing
sizeof(AVPacket) from the API which means you cannot stack-allocate
AVPacket anymore. However, that is something we take advantage of
because we use short-lived AVPackets to bridge from native mpv packets
in our main decoding paths.

We don't think that switching these to `av_packet_alloc` is desirable,
given the cost of heap allocation, so this change takes a different
approach - allocating a single packet in the relevant context and
reusing it over and over.

That's fairly straight-forward, with the main caveat being that
re-initialising the packet is unintuitive. There is no function that
does exactly what we need (what `av_init_packet` did). The closest is
`av_packet_unref`, which additionally frees buffers and side-data.
However, we don't copy those things - we just assign them in from our
own packet, so we have to explicitly clear the pointers before calling
`av_packet_unref`. But at least we can make a wrapper function for
that.

The weirdest part of the change is the handling of the vtt subtitle
conversion. This requires two packets, so I had to pre-allocate two in
the context struct. That sounds excessive, but if allocating the
primary packet is too expensive, then allocating the secondary one for
vtt subtitles must also be too expensive.

This change is not conditional as heap allocated AVPackets were
available for years and years before the deprecation.
2022-12-03 14:44:18 -08:00

326 lines
9.3 KiB
C

/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or
* modify it under the terms of the GNU 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 <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <stdbool.h>
#include <assert.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include <libavutil/common.h>
#include <libavutil/intreadwrite.h>
#include "config.h"
#include "mpv_talloc.h"
#include "audio/aframe.h"
#include "audio/chmap_avchannel.h"
#include "audio/fmt-conversion.h"
#include "common/av_common.h"
#include "common/codecs.h"
#include "common/global.h"
#include "common/msg.h"
#include "demux/packet.h"
#include "demux/stheader.h"
#include "filters/f_decoder_wrapper.h"
#include "filters/filter_internal.h"
#include "options/m_config.h"
#include "options/options.h"
struct priv {
AVCodecContext *avctx;
AVFrame *avframe;
AVPacket *avpkt;
struct mp_chmap force_channel_map;
uint32_t skip_samples, trim_samples;
bool preroll_done;
double next_pts;
AVRational codec_timebase;
struct lavc_state state;
struct mp_decoder public;
};
#define OPT_BASE_STRUCT struct ad_lavc_params
struct ad_lavc_params {
float ac3drc;
int downmix;
int threads;
char **avopts;
};
const struct m_sub_options ad_lavc_conf = {
.opts = (const m_option_t[]) {
{"ac3drc", OPT_FLOAT(ac3drc), M_RANGE(0, 6)},
{"downmix", OPT_FLAG(downmix)},
{"threads", OPT_INT(threads), M_RANGE(0, 16)},
{"o", OPT_KEYVALUELIST(avopts)},
{0}
},
.size = sizeof(struct ad_lavc_params),
.defaults = &(const struct ad_lavc_params){
.ac3drc = 0,
.downmix = 0,
.threads = 1,
},
};
static bool init(struct mp_filter *da, struct mp_codec_params *codec,
const char *decoder)
{
struct priv *ctx = da->priv;
struct MPOpts *mpopts = mp_get_config_group(ctx, da->global, &mp_opt_root);
struct ad_lavc_params *opts =
mp_get_config_group(ctx, da->global, &ad_lavc_conf);
AVCodecContext *lavc_context;
const AVCodec *lavc_codec;
ctx->codec_timebase = mp_get_codec_timebase(codec);
if (codec->force_channels)
ctx->force_channel_map = codec->channels;
lavc_codec = avcodec_find_decoder_by_name(decoder);
if (!lavc_codec) {
MP_ERR(da, "Cannot find codec '%s' in libavcodec...\n", decoder);
return false;
}
lavc_context = avcodec_alloc_context3(lavc_codec);
ctx->avctx = lavc_context;
ctx->avframe = av_frame_alloc();
ctx->avpkt = av_packet_alloc();
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id;
lavc_context->pkt_timebase = ctx->codec_timebase;
if (opts->downmix && mpopts->audio_output_channels.num_chmaps == 1) {
const struct mp_chmap *requested_layout =
&mpopts->audio_output_channels.chmaps[0];
#if !HAVE_AV_CHANNEL_LAYOUT
lavc_context->request_channel_layout =
mp_chmap_to_lavc(requested_layout);
#else
AVChannelLayout av_layout = { 0 };
mp_chmap_to_av_layout(&av_layout, requested_layout);
// Always try to set requested output layout - currently only something
// supported by AC3, MLP/TrueHD, DTS and the fdk-aac wrapper.
av_opt_set_chlayout(lavc_context, "downmix", &av_layout,
AV_OPT_SEARCH_CHILDREN);
av_channel_layout_uninit(&av_layout);
#endif
}
// Always try to set - option only exists for AC3 at the moment
av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
AV_OPT_SEARCH_CHILDREN);
// Let decoder add AV_FRAME_DATA_SKIP_SAMPLES.
av_opt_set(lavc_context, "flags2", "+skip_manual", AV_OPT_SEARCH_CHILDREN);
mp_set_avopts(da->log, lavc_context, opts->avopts);
if (mp_set_avctx_codec_headers(lavc_context, codec) < 0) {
MP_ERR(da, "Could not set decoder parameters.\n");
return false;
}
mp_set_avcodec_threads(da->log, lavc_context, opts->threads);
/* open it */
if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
MP_ERR(da, "Could not open codec.\n");
return false;
}
ctx->next_pts = MP_NOPTS_VALUE;
return true;
}
static void destroy(struct mp_filter *da)
{
struct priv *ctx = da->priv;
avcodec_free_context(&ctx->avctx);
av_frame_free(&ctx->avframe);
mp_free_av_packet(&ctx->avpkt);
}
static void reset(struct mp_filter *da)
{
struct priv *ctx = da->priv;
avcodec_flush_buffers(ctx->avctx);
ctx->skip_samples = 0;
ctx->trim_samples = 0;
ctx->preroll_done = false;
ctx->next_pts = MP_NOPTS_VALUE;
ctx->state = (struct lavc_state){0};
}
static int send_packet(struct mp_filter *da, struct demux_packet *mpkt)
{
struct priv *priv = da->priv;
AVCodecContext *avctx = priv->avctx;
// If the decoder discards the timestamp for some reason, we use the
// interpolated PTS. Initialize it so that it works for the initial
// packet as well.
if (mpkt && priv->next_pts == MP_NOPTS_VALUE)
priv->next_pts = mpkt->pts;
mp_set_av_packet(priv->avpkt, mpkt, &priv->codec_timebase);
int ret = avcodec_send_packet(avctx, mpkt ? priv->avpkt : NULL);
if (ret < 0)
MP_ERR(da, "Error decoding audio.\n");
return ret;
}
static int receive_frame(struct mp_filter *da, struct mp_frame *out)
{
struct priv *priv = da->priv;
AVCodecContext *avctx = priv->avctx;
int ret = avcodec_receive_frame(avctx, priv->avframe);
if (ret == AVERROR_EOF) {
// If flushing was initialized earlier and has ended now, make it start
// over in case we get new packets at some point in the future.
// (Dont' reset the filter itself, we want to keep other state.)
avcodec_flush_buffers(priv->avctx);
return ret;
} else if (ret < 0 && ret != AVERROR(EAGAIN)) {
MP_ERR(da, "Error decoding audio.\n");
}
if (priv->avframe->flags & AV_FRAME_FLAG_DISCARD)
av_frame_unref(priv->avframe);
if (!priv->avframe->buf[0])
return ret;
double out_pts = mp_pts_from_av(priv->avframe->pts, &priv->codec_timebase);
struct mp_aframe *mpframe = mp_aframe_from_avframe(priv->avframe);
if (!mpframe) {
MP_ERR(da, "Converting libavcodec frame to mpv frame failed.\n");
return ret;
}
if (priv->force_channel_map.num)
mp_aframe_set_chmap(mpframe, &priv->force_channel_map);
if (out_pts == MP_NOPTS_VALUE)
out_pts = priv->next_pts;
mp_aframe_set_pts(mpframe, out_pts);
priv->next_pts = mp_aframe_end_pts(mpframe);
AVFrameSideData *sd =
av_frame_get_side_data(priv->avframe, AV_FRAME_DATA_SKIP_SAMPLES);
if (sd && sd->size >= 10) {
char *d = sd->data;
priv->skip_samples += AV_RL32(d + 0);
priv->trim_samples += AV_RL32(d + 4);
}
if (!priv->preroll_done) {
// Skip only if this isn't already handled by AV_FRAME_DATA_SKIP_SAMPLES.
if (!priv->skip_samples)
priv->skip_samples = avctx->delay;
priv->preroll_done = true;
}
uint32_t skip = MPMIN(priv->skip_samples, mp_aframe_get_size(mpframe));
if (skip) {
mp_aframe_skip_samples(mpframe, skip);
priv->skip_samples -= skip;
}
uint32_t trim = MPMIN(priv->trim_samples, mp_aframe_get_size(mpframe));
if (trim) {
mp_aframe_set_size(mpframe, mp_aframe_get_size(mpframe) - trim);
priv->trim_samples -= trim;
}
// Strip possibly bogus float values like Infinity, NaN, denormalized
mp_aframe_sanitize_float(mpframe);
if (mp_aframe_get_size(mpframe) > 0) {
*out = MAKE_FRAME(MP_FRAME_AUDIO, mpframe);
} else {
talloc_free(mpframe);
}
av_frame_unref(priv->avframe);
return ret;
}
static void process(struct mp_filter *ad)
{
struct priv *priv = ad->priv;
lavc_process(ad, &priv->state, send_packet, receive_frame);
}
static const struct mp_filter_info ad_lavc_filter = {
.name = "ad_lavc",
.priv_size = sizeof(struct priv),
.process = process,
.reset = reset,
.destroy = destroy,
};
static struct mp_decoder *create(struct mp_filter *parent,
struct mp_codec_params *codec,
const char *decoder)
{
struct mp_filter *da = mp_filter_create(parent, &ad_lavc_filter);
if (!da)
return NULL;
mp_filter_add_pin(da, MP_PIN_IN, "in");
mp_filter_add_pin(da, MP_PIN_OUT, "out");
da->log = mp_log_new(da, parent->log, NULL);
struct priv *priv = da->priv;
priv->public.f = da;
if (!init(da, codec, decoder)) {
talloc_free(da);
return NULL;
}
return &priv->public;
}
static void add_decoders(struct mp_decoder_list *list)
{
mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO);
}
const struct mp_decoder_fns ad_lavc = {
.create = create,
.add_decoders = add_decoders,
};