/* * 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 . */ #include #include #include #include #include #include #include #include #include #include "mpv_talloc.h" #include "audio/aframe.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/options.h" struct priv { AVCodecContext *avctx; AVFrame *avframe; struct mp_chmap force_channel_map; uint32_t skip_samples, trim_samples; bool preroll_done; double next_pts; AVRational codec_timebase; bool eof_returned; 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[]) { 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 bool init(struct mp_filter *da, struct mp_codec_params *codec, const char *decoder) { struct priv *ctx = da->priv; struct MPOpts *mpopts = da->global->opts; struct ad_lavc_params *opts = mpopts->ad_lavc_params; AVCodecContext *lavc_context; 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(); 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, 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); } 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->eof_returned = false; } static bool 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; 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 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 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.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); #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 = MAKE_FRAME(MP_FRAME_AUDIO, mpframe); av_frame_unref(priv->avframe); return true; } static void process(struct mp_filter *ad) { struct priv *priv = ad->priv; lavc_process(ad, &priv->eof_returned, 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, };