mirror of
https://github.com/mpv-player/mpv
synced 2024-12-21 22:30:22 +00:00
5fd8a1e04c
This rewrites the audio decode loop to some degree. Audio filters don't do refcounted frames yet, so af.c contains a hacky "emulation". Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of estimating how much audio we need to filter, we always filter full frames. Maybe this should be adjusted later: in case filtering increases the volume of the audio data, we should try not to buffer too much filter output by reducing the input that is fed at once. For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it doesn't seem worth the trouble.
294 lines
8.4 KiB
C
294 lines
8.4 KiB
C
/*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer 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 General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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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 "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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struct demux_packet *packet;
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uint32_t skip_samples;
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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, 2),
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OPT_FLAG("downmix", downmix, 0),
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OPT_INTRANGE("threads", threads, 0, 1, 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 = 1.,
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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 sh_stream *sh = da->header;
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struct sh_audio *sh_audio = sh->audio;
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struct priv *ctx = talloc_zero(NULL, struct priv);
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da->priv = ctx;
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ctx->force_channel_map = sh_audio->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->refcounted_frames = 1;
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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 (opts->downmix) {
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lavc_context->request_channel_layout =
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mp_chmap_to_lavc(&mpopts->audio_output_channels);
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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 HAVE_AVFRAME_SKIP_SAMPLES
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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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lavc_context->codec_tag = sh->format;
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lavc_context->sample_rate = sh_audio->samplerate;
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lavc_context->bit_rate = sh_audio->bitrate;
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lavc_context->block_align = sh_audio->block_align;
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lavc_context->bits_per_coded_sample = sh_audio->bits_per_coded_sample;
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lavc_context->channels = sh_audio->channels.num;
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if (!mp_chmap_is_unknown(&sh_audio->channels))
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lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
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// demux_mkv
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if (sh_audio->codecdata_len && sh_audio->codecdata &&
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!lavc_context->extradata) {
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mp_lavc_set_extradata(lavc_context, sh_audio->codecdata,
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sh_audio->codecdata_len);
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}
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if (sh->lav_headers)
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mp_copy_lav_codec_headers(lavc_context, sh->lav_headers);
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mp_set_avcodec_threads(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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if (lavc_context->bit_rate != 0)
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da->bitrate = lavc_context->bit_rate;
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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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talloc_free(ctx->packet);
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ctx->packet = NULL;
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ctx->skip_samples = 0;
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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 int decode_packet(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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struct demux_packet *mpkt = priv->packet;
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if (!mpkt) {
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if (demux_read_packet_async(da->header, &mpkt) == 0)
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return AD_WAIT;
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}
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priv->packet = talloc_steal(priv, mpkt);
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int in_len = mpkt ? mpkt->len : 0;
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AVPacket pkt;
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mp_set_av_packet(&pkt, mpkt, NULL);
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// If we don't have a PTS yet, use the first packet PTS we can get.
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if (da->pts == MP_NOPTS_VALUE && mpkt && mpkt->pts != MP_NOPTS_VALUE) {
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da->pts = mpkt->pts;
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da->pts_offset = 0;
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}
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int got_frame = 0;
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av_frame_unref(priv->avframe);
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int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
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if (mpkt) {
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// At least "shorten" decodes sub-frames, instead of the whole packet.
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// At least "mpc8" can return 0 and wants the packet again next time.
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if (ret >= 0) {
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ret = FFMIN(ret, mpkt->len); // sanity check against decoder overreads
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mpkt->buffer += ret;
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mpkt->len -= ret;
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mpkt->pts = MP_NOPTS_VALUE; // don't reset PTS next time
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}
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if (mpkt->len == 0 || ret < 0) {
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talloc_free(mpkt);
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priv->packet = NULL;
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}
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// LATM may need many packets to find mux info
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if (ret == AVERROR(EAGAIN))
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return AD_OK;
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}
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if (ret < 0) {
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MP_ERR(da, "Error decoding audio.\n");
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return AD_ERR;
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}
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if (!got_frame)
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return mpkt ? AD_OK : AD_EOF;
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double out_pts = mp_pts_from_av(priv->avframe->pkt_pts, NULL);
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if (out_pts != MP_NOPTS_VALUE) {
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da->pts = out_pts;
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da->pts_offset = 0;
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}
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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 AD_ERR;
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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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struct sh_audio *sh_audio = da->header->audio;
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if (lavc_chmap.num == sh_audio->channels.num)
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lavc_chmap = sh_audio->channels;
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}
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mp_audio_set_channels(mpframe, &lavc_chmap);
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#if HAVE_AVFRAME_SKIP_SAMPLES
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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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uint32_t pad = AV_RL32(d + 4);
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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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da->pts_offset += skip;
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priv->skip_samples -= skip;
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}
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if (pad <= mpframe->samples)
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mpframe->samples -= pad;
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}
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#endif
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*out = mpframe;
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av_frame_unref(priv->avframe);
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MP_DBG(da, "Decoded %d -> %d samples\n", in_len, mpframe->samples);
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return 0;
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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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.decode_packet = decode_packet,
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};
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