mirror of https://github.com/mpv-player/mpv
296 lines
8.2 KiB
C
296 lines
8.2 KiB
C
/*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <stdbool.h>
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#include <assert.h>
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#include <libavcodec/avcodec.h>
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#include <libavutil/opt.h>
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#include <libavutil/common.h>
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#include <libavutil/intreadwrite.h>
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#include "mpv_talloc.h"
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#include "config.h"
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#include "common/av_common.h"
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#include "common/codecs.h"
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#include "common/msg.h"
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#include "options/options.h"
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#include "ad.h"
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#include "audio/fmt-conversion.h"
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struct priv {
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AVCodecContext *avctx;
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AVFrame *avframe;
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struct mp_audio frame;
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bool force_channel_map;
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uint32_t skip_samples, trim_samples;
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bool preroll_done;
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double next_pts;
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AVRational codec_timebase;
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};
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static void uninit(struct dec_audio *da);
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#define OPT_BASE_STRUCT struct ad_lavc_params
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struct ad_lavc_params {
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float ac3drc;
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int downmix;
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int threads;
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char **avopts;
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};
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const struct m_sub_options ad_lavc_conf = {
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.opts = (const m_option_t[]) {
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OPT_FLOATRANGE("ac3drc", ac3drc, 0, 0, 6),
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OPT_FLAG("downmix", downmix, 0),
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OPT_INTRANGE("threads", threads, 0, 0, 16),
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OPT_KEYVALUELIST("o", avopts, 0),
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{0}
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},
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.size = sizeof(struct ad_lavc_params),
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.defaults = &(const struct ad_lavc_params){
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.ac3drc = 0,
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.downmix = 1,
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.threads = 1,
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},
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};
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static int init(struct dec_audio *da, const char *decoder)
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{
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struct MPOpts *mpopts = da->opts;
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struct ad_lavc_params *opts = mpopts->ad_lavc_params;
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AVCodecContext *lavc_context;
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AVCodec *lavc_codec;
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struct mp_codec_params *c = da->codec;
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struct priv *ctx = talloc_zero(NULL, struct priv);
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da->priv = ctx;
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ctx->codec_timebase = mp_get_codec_timebase(da->codec);
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ctx->force_channel_map = c->force_channels;
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lavc_codec = avcodec_find_decoder_by_name(decoder);
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if (!lavc_codec) {
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MP_ERR(da, "Cannot find codec '%s' in libavcodec...\n", decoder);
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uninit(da);
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return 0;
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}
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lavc_context = avcodec_alloc_context3(lavc_codec);
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ctx->avctx = lavc_context;
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ctx->avframe = av_frame_alloc();
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lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
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lavc_context->codec_id = lavc_codec->id;
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#if LIBAVCODEC_VERSION_MICRO >= 100
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lavc_context->pkt_timebase = ctx->codec_timebase;
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#endif
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if (opts->downmix && mpopts->audio_output_channels.num_chmaps == 1) {
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lavc_context->request_channel_layout =
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mp_chmap_to_lavc(&mpopts->audio_output_channels.chmaps[0]);
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}
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// Always try to set - option only exists for AC3 at the moment
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av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
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AV_OPT_SEARCH_CHILDREN);
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#if LIBAVCODEC_VERSION_MICRO >= 100
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// Let decoder add AV_FRAME_DATA_SKIP_SAMPLES.
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av_opt_set(lavc_context, "flags2", "+skip_manual", AV_OPT_SEARCH_CHILDREN);
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#endif
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mp_set_avopts(da->log, lavc_context, opts->avopts);
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if (mp_set_avctx_codec_headers(lavc_context, c) < 0) {
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MP_ERR(da, "Could not set decoder parameters.\n");
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uninit(da);
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return 0;
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}
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mp_set_avcodec_threads(da->log, lavc_context, opts->threads);
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/* open it */
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if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
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MP_ERR(da, "Could not open codec.\n");
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uninit(da);
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return 0;
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}
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ctx->next_pts = MP_NOPTS_VALUE;
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return 1;
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}
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static void uninit(struct dec_audio *da)
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{
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struct priv *ctx = da->priv;
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if (!ctx)
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return;
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AVCodecContext *lavc_context = ctx->avctx;
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if (lavc_context) {
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if (avcodec_close(lavc_context) < 0)
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MP_ERR(da, "Could not close codec.\n");
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av_freep(&lavc_context->extradata);
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av_freep(&lavc_context);
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}
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av_frame_free(&ctx->avframe);
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}
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static int control(struct dec_audio *da, int cmd, void *arg)
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{
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struct priv *ctx = da->priv;
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switch (cmd) {
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case ADCTRL_RESET:
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avcodec_flush_buffers(ctx->avctx);
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ctx->skip_samples = 0;
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ctx->trim_samples = 0;
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ctx->preroll_done = false;
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ctx->next_pts = MP_NOPTS_VALUE;
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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static bool send_packet(struct dec_audio *da, struct demux_packet *mpkt)
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{
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struct priv *priv = da->priv;
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AVCodecContext *avctx = priv->avctx;
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// If the decoder discards the timestamp for some reason, we use the
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// interpolated PTS. Initialize it so that it works for the initial
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// packet as well.
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if (mpkt && priv->next_pts == MP_NOPTS_VALUE)
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priv->next_pts = mpkt->pts;
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AVPacket pkt;
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mp_set_av_packet(&pkt, mpkt, &priv->codec_timebase);
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int ret = avcodec_send_packet(avctx, mpkt ? &pkt : NULL);
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if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
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return false;
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if (ret < 0)
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MP_ERR(da, "Error decoding audio.\n");
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return true;
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}
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static bool receive_frame(struct dec_audio *da, struct mp_audio **out)
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{
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struct priv *priv = da->priv;
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AVCodecContext *avctx = priv->avctx;
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int ret = avcodec_receive_frame(avctx, priv->avframe);
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if (ret == AVERROR_EOF) {
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// If flushing was initialized earlier and has ended now, make it start
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// over in case we get new packets at some point in the future.
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control(da, ADCTRL_RESET, NULL);
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return false;
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} else if (ret < 0 && ret != AVERROR(EAGAIN)) {
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MP_ERR(da, "Error decoding audio.\n");
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}
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#if LIBAVCODEC_VERSION_MICRO >= 100
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if (priv->avframe->flags & AV_FRAME_FLAG_DISCARD)
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av_frame_unref(priv->avframe);
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#endif
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if (!priv->avframe->buf[0])
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return true;
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double out_pts = mp_pts_from_av(priv->avframe->pts, &priv->codec_timebase);
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struct mp_audio *mpframe = mp_audio_from_avframe(priv->avframe);
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if (!mpframe)
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return true;
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struct mp_chmap lavc_chmap = mpframe->channels;
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if (lavc_chmap.num != avctx->channels)
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mp_chmap_from_channels(&lavc_chmap, avctx->channels);
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if (priv->force_channel_map) {
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if (lavc_chmap.num == da->codec->channels.num)
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lavc_chmap = da->codec->channels;
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}
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mp_audio_set_channels(mpframe, &lavc_chmap);
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mpframe->pts = out_pts;
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if (mpframe->pts == MP_NOPTS_VALUE)
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mpframe->pts = priv->next_pts;
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if (mpframe->pts != MP_NOPTS_VALUE)
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priv->next_pts = mpframe->pts + mpframe->samples / (double)mpframe->rate;
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#if LIBAVCODEC_VERSION_MICRO >= 100
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AVFrameSideData *sd =
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av_frame_get_side_data(priv->avframe, AV_FRAME_DATA_SKIP_SAMPLES);
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if (sd && sd->size >= 10) {
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char *d = sd->data;
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priv->skip_samples += AV_RL32(d + 0);
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priv->trim_samples += AV_RL32(d + 4);
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}
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#endif
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if (!priv->preroll_done) {
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// Skip only if this isn't already handled by AV_FRAME_DATA_SKIP_SAMPLES.
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if (!priv->skip_samples)
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priv->skip_samples = avctx->delay;
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priv->preroll_done = true;
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}
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uint32_t skip = MPMIN(priv->skip_samples, mpframe->samples);
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if (skip) {
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mp_audio_skip_samples(mpframe, skip);
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priv->skip_samples -= skip;
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}
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uint32_t trim = MPMIN(priv->trim_samples, mpframe->samples);
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if (trim) {
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mpframe->samples -= trim;
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priv->trim_samples -= trim;
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}
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*out = mpframe;
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av_frame_unref(priv->avframe);
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MP_DBG(da, "Decoded %d samples\n", mpframe->samples);
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return true;
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}
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static void add_decoders(struct mp_decoder_list *list)
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{
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mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO);
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}
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const struct ad_functions ad_lavc = {
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.name = "lavc",
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.add_decoders = add_decoders,
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.init = init,
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.uninit = uninit,
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.control = control,
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.send_packet = send_packet,
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.receive_frame = receive_frame,
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};
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