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mirror of https://github.com/mpv-player/mpv.git synced 2024-09-20 03:52:22 +02:00
mpv/audio/decode/ad_lavc.c
wm4 1f593beeb4 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 21:10:54 +02:00

281 lines
7.6 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 "mpv_talloc.h"
#include "config.h"
#include "common/av_common.h"
#include "common/codecs.h"
#include "common/msg.h"
#include "options/options.h"
#include "ad.h"
#include "audio/fmt-conversion.h"
struct priv {
AVCodecContext *avctx;
AVFrame *avframe;
bool force_channel_map;
uint32_t skip_samples, trim_samples;
bool preroll_done;
double next_pts;
AVRational codec_timebase;
};
static void uninit(struct dec_audio *da);
#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[]) {
OPT_FLOATRANGE("ac3drc", ac3drc, 0, 0, 6),
OPT_FLAG("downmix", downmix, 0),
OPT_INTRANGE("threads", threads, 0, 0, 16),
OPT_KEYVALUELIST("o", avopts, 0),
{0}
},
.size = sizeof(struct ad_lavc_params),
.defaults = &(const struct ad_lavc_params){
.ac3drc = 0,
.downmix = 1,
.threads = 1,
},
};
static int init(struct dec_audio *da, const char *decoder)
{
struct MPOpts *mpopts = da->opts;
struct ad_lavc_params *opts = mpopts->ad_lavc_params;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
struct mp_codec_params *c = da->codec;
struct priv *ctx = talloc_zero(NULL, struct priv);
da->priv = ctx;
ctx->codec_timebase = mp_get_codec_timebase(da->codec);
ctx->force_channel_map = c->force_channels;
lavc_codec = avcodec_find_decoder_by_name(decoder);
if (!lavc_codec) {
MP_ERR(da, "Cannot find codec '%s' in libavcodec...\n", decoder);
uninit(da);
return 0;
}
lavc_context = avcodec_alloc_context3(lavc_codec);
ctx->avctx = lavc_context;
ctx->avframe = av_frame_alloc();
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id;
#if LIBAVCODEC_VERSION_MICRO >= 100
lavc_context->pkt_timebase = ctx->codec_timebase;
#endif
if (opts->downmix && mpopts->audio_output_channels.num_chmaps == 1) {
lavc_context->request_channel_layout =
mp_chmap_to_lavc(&mpopts->audio_output_channels.chmaps[0]);
}
// 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);
#if LIBAVCODEC_VERSION_MICRO >= 100
// Let decoder add AV_FRAME_DATA_SKIP_SAMPLES.
av_opt_set(lavc_context, "flags2", "+skip_manual", AV_OPT_SEARCH_CHILDREN);
#endif
mp_set_avopts(da->log, lavc_context, opts->avopts);
if (mp_set_avctx_codec_headers(lavc_context, c) < 0) {
MP_ERR(da, "Could not set decoder parameters.\n");
uninit(da);
return 0;
}
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");
uninit(da);
return 0;
}
ctx->next_pts = MP_NOPTS_VALUE;
return 1;
}
static void uninit(struct dec_audio *da)
{
struct priv *ctx = da->priv;
if (!ctx)
return;
avcodec_free_context(&ctx->avctx);
av_frame_free(&ctx->avframe);
}
static int control(struct dec_audio *da, int cmd, void *arg)
{
struct priv *ctx = da->priv;
switch (cmd) {
case ADCTRL_RESET:
avcodec_flush_buffers(ctx->avctx);
ctx->skip_samples = 0;
ctx->trim_samples = 0;
ctx->preroll_done = false;
ctx->next_pts = MP_NOPTS_VALUE;
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static bool send_packet(struct dec_audio *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;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, &priv->codec_timebase);
int ret = avcodec_send_packet(avctx, mpkt ? &pkt : NULL);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return false;
if (ret < 0)
MP_ERR(da, "Error decoding audio.\n");
return true;
}
static bool receive_frame(struct dec_audio *da, struct mp_aframe **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.
control(da, ADCTRL_RESET, NULL);
return false;
} else if (ret < 0 && ret != AVERROR(EAGAIN)) {
MP_ERR(da, "Error decoding audio.\n");
}
#if LIBAVCODEC_VERSION_MICRO >= 100
if (priv->avframe->flags & AV_FRAME_FLAG_DISCARD)
av_frame_unref(priv->avframe);
#endif
if (!priv->avframe->buf[0])
return true;
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)
return true;
if (priv->force_channel_map)
mp_aframe_set_chmap(mpframe, &da->codec->channels);
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);
#if LIBAVCODEC_VERSION_MICRO >= 100
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);
}
#endif
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;
}
*out = mpframe;
av_frame_unref(priv->avframe);
return true;
}
static void add_decoders(struct mp_decoder_list *list)
{
mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO);
}
const struct ad_functions ad_lavc = {
.name = "lavc",
.add_decoders = add_decoders,
.init = init,
.uninit = uninit,
.control = control,
.send_packet = send_packet,
.receive_frame = receive_frame,
};