mirror of https://github.com/mpv-player/mpv
414 lines
13 KiB
C
414 lines
13 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 "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 "common/av_opts.h"
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#include "ad.h"
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#include "audio/fmt-conversion.h"
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#include "compat/libav.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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};
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static void uninit(struct dec_audio *da);
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static int decode_new_packet(struct dec_audio *da);
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#define OPT_BASE_STRUCT struct MPOpts
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const m_option_t ad_lavc_decode_opts_conf[] = {
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OPT_FLOATRANGE("ac3drc", ad_lavc_param.ac3drc, 0, 0, 2),
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OPT_FLAG("downmix", ad_lavc_param.downmix, 0),
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OPT_INTRANGE("threads", ad_lavc_param.threads, 0, 1, 16),
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OPT_STRING("o", ad_lavc_param.avopt, 0),
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{0}
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};
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struct pcm_map
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{
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int tag;
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const char *codecs[6]; // {any, 1byte, 2bytes, 3bytes, 4bytes, 8bytes}
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};
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// NOTE: these are needed to make rawaudio with demux_mkv work.
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static const struct pcm_map tag_map[] = {
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// Microsoft PCM
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{0x0, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
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{0x1, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
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// MS PCM, Extended
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{0xfffe, {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
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// IEEE float
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{0x3, {"pcm_f32le", [5] = "pcm_f64le"}},
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// 'raw '
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{0x20776172, {"pcm_s16be", [1] = "pcm_u8"}},
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// 'twos', used by demux_mkv.c internally
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{MKTAG('t', 'w', 'o', 's'),
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{NULL, "pcm_s8", "pcm_s16be", "pcm_s24be", "pcm_s32be"}},
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{-1},
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};
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// For demux_rawaudio.c; needed because ffmpeg doesn't have these sample
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// formats natively.
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static const struct pcm_map af_map[] = {
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{AF_FORMAT_U8, {"pcm_u8"}},
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{AF_FORMAT_S8, {"pcm_u8"}},
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{AF_FORMAT_U16_LE, {"pcm_u16le"}},
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{AF_FORMAT_U16_BE, {"pcm_u16be"}},
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{AF_FORMAT_S16_LE, {"pcm_s16le"}},
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{AF_FORMAT_S16_BE, {"pcm_s16be"}},
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{AF_FORMAT_U24_LE, {"pcm_u24le"}},
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{AF_FORMAT_U24_BE, {"pcm_u24be"}},
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{AF_FORMAT_S24_LE, {"pcm_s24le"}},
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{AF_FORMAT_S24_BE, {"pcm_s24be"}},
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{AF_FORMAT_U32_LE, {"pcm_u32le"}},
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{AF_FORMAT_U32_BE, {"pcm_u32be"}},
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{AF_FORMAT_S32_LE, {"pcm_s32le"}},
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{AF_FORMAT_S32_BE, {"pcm_s32be"}},
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{AF_FORMAT_FLOAT_LE, {"pcm_f32le"}},
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{AF_FORMAT_FLOAT_BE, {"pcm_f32be"}},
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{AF_FORMAT_DOUBLE_LE, {"pcm_f64le"}},
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{AF_FORMAT_DOUBLE_BE, {"pcm_f64be"}},
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{-1},
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};
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static const char *find_pcm_decoder(const struct pcm_map *map, int format,
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int bits_per_sample)
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{
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int bytes = (bits_per_sample + 7) / 8;
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if (bytes == 8)
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bytes = 5; // 64 bit entry
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for (int n = 0; map[n].tag != -1; n++) {
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const struct pcm_map *entry = &map[n];
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if (entry->tag == format) {
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const char *dec = NULL;
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if (bytes >= 1 && bytes <= 5)
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dec = entry->codecs[bytes];
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if (!dec)
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dec = entry->codecs[0];
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if (dec)
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return dec;
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}
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}
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return NULL;
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}
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static int setup_format(struct dec_audio *da)
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{
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struct priv *priv = da->priv;
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AVCodecContext *lavc_context = priv->avctx;
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struct sh_audio *sh_audio = da->header->audio;
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// Note: invalid parameters are rejected by dec_audio.c
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int fmt = lavc_context->sample_fmt;
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mp_audio_set_format(&da->decoded, af_from_avformat(fmt));
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if (!da->decoded.format)
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MP_FATAL(da, "unsupported lavc format %s", av_get_sample_fmt_name(fmt));
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da->decoded.rate = lavc_context->sample_rate;
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if (!da->decoded.rate && sh_audio->wf) {
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// If not set, try container samplerate.
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// (Maybe this can't happen, and it's an artifact from the past.)
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da->decoded.rate = sh_audio->wf->nSamplesPerSec;
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MP_WARN(da, "using container rate.\n");
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}
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struct mp_chmap lavc_chmap;
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mp_chmap_from_lavc(&lavc_chmap, lavc_context->channel_layout);
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// No channel layout or layout disagrees with channel count
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if (lavc_chmap.num != lavc_context->channels)
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mp_chmap_from_channels(&lavc_chmap, lavc_context->channels);
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if (priv->force_channel_map) {
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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(&da->decoded, &lavc_chmap);
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return 0;
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}
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static void set_from_wf(AVCodecContext *avctx, MP_WAVEFORMATEX *wf)
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{
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avctx->channels = wf->nChannels;
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avctx->sample_rate = wf->nSamplesPerSec;
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avctx->bit_rate = wf->nAvgBytesPerSec * 8;
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avctx->block_align = wf->nBlockAlign;
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avctx->bits_per_coded_sample = wf->wBitsPerSample;
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if (wf->cbSize > 0)
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mp_lavc_set_extradata(avctx, wf + 1, wf->cbSize);
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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_param *opts = &mpopts->ad_lavc_param;
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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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if (sh_audio->wf && strcmp(decoder, "pcm") == 0) {
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decoder = find_pcm_decoder(tag_map, sh->format,
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sh_audio->wf->wBitsPerSample);
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} else if (sh_audio->wf && strcmp(decoder, "mp-pcm") == 0) {
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decoder = find_pcm_decoder(af_map, sh->format, 0);
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ctx->force_channel_map = true;
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}
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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 (opts->avopt) {
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if (parse_avopts(lavc_context, opts->avopt) < 0) {
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MP_ERR(da, "setting AVOptions '%s' failed.\n", opts->avopt);
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uninit(da);
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return 0;
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}
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}
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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->i_bps * 8;
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lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);
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if (sh_audio->wf)
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set_from_wf(lavc_context, sh_audio->wf);
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// demux_mkv, demux_mpg
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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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MP_VERBOSE(da, "INFO: libavcodec \"%s\" init OK!\n",
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lavc_codec->name);
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// Decode at least 1 sample: (to get header filled)
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for (int tries = 1; ; tries++) {
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int x = decode_new_packet(da);
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if (x >= 0 && ctx->frame.samples > 0) {
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MP_VERBOSE(da, "Initial decode succeeded after %d packets.\n", tries);
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break;
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}
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if (tries >= 50) {
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MP_ERR(da, "initial decode failed\n");
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uninit(da);
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return 0;
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}
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}
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da->i_bps = lavc_context->bit_rate / 8;
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if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
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da->i_bps = sh_audio->wf->nAvgBytesPerSec;
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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->frame.samples = 0;
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talloc_free(ctx->packet);
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ctx->packet = NULL;
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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_new_packet(struct dec_audio *da)
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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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priv->frame.samples = 0;
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struct demux_packet *mpkt = priv->packet;
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if (!mpkt)
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mpkt = demux_read_packet(da->header);
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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 0;
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}
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if (ret < 0) {
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MP_VERBOSE(da, "lavc_audio: error\n");
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return -1;
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}
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if (!got_frame)
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return mpkt ? 0 : -1; // -1: eof
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if (setup_format(da) < 0)
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return -1;
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priv->frame.samples = priv->avframe->nb_samples;
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mp_audio_copy_config(&priv->frame, &da->decoded);
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for (int n = 0; n < priv->frame.num_planes; n++)
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priv->frame.planes[n] = priv->avframe->data[n];
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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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MP_DBG(da, "Decoded %d -> %d samples\n", in_len,
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priv->frame.samples);
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return 0;
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}
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static int decode_audio(struct dec_audio *da, struct mp_audio *buffer, int maxlen)
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{
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struct priv *priv = da->priv;
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if (!priv->frame.samples) {
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if (decode_new_packet(da) < 0)
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return -1;
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}
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if (!mp_audio_config_equals(buffer, &priv->frame))
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return 0;
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buffer->samples = MPMIN(priv->frame.samples, maxlen);
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mp_audio_copy(buffer, 0, &priv->frame, 0, buffer->samples);
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mp_audio_skip_samples(&priv->frame, buffer->samples);
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da->pts_offset += buffer->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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mp_add_decoder(list, "lavc", "pcm", "pcm", "Raw PCM");
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mp_add_decoder(list, "lavc", "mp-pcm", "mp-pcm", "Raw PCM");
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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_audio = decode_audio,
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};
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