mirror of
https://github.com/mpv-player/mpv
synced 2024-12-25 16:33:02 +00:00
d8882bbfb7
This affects 64 bit floats and big endian integer PCM variants (basically crap nobody uses). Possibly not all MS-muxed files work, but I couldn't get or produce any samples. Remove a bunch of format tags that are not needed anymore. Most of these were used by demux_mov, which is long gone. Repurpose/abuse 'twos' as mpv-internal tag for dealing with the PCM variants mentioned above.
470 lines
15 KiB
C
470 lines
15 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 "mpvcore/av_common.h"
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#include "mpvcore/codecs.h"
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#include "mpvcore/mp_msg.h"
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#include "mpvcore/options.h"
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#include "mpvcore/av_opts.h"
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#include "ad.h"
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#include "audio/reorder_ch.h"
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#include "audio/fmt-conversion.h"
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#include "compat/mpbswap.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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uint8_t *output;
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uint8_t *output_packed; // used by deplanarize to store packed audio samples
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int output_left;
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int unitsize;
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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(sh_audio_t *sh);
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static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen);
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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_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 preinit(sh_audio_t *sh)
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{
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return 1;
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}
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/* Prefer playing audio with the samplerate given in container data
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* if available, but take number the number of channels and sample format
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* from the codec, since if the codec isn't using the correct values for
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* those everything breaks anyway.
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*/
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static int setup_format(sh_audio_t *sh_audio,
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const AVCodecContext *lavc_context)
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{
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struct priv *priv = sh_audio->context;
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int sample_format =
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af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
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int samplerate = lavc_context->sample_rate;
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// If not set, try container samplerate
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if (!samplerate && sh_audio->wf) {
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samplerate = sh_audio->wf->nSamplesPerSec;
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mp_tmsg(MSGT_DECAUDIO, MSGL_V, "ad_lavc: 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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if (!mp_chmap_equals(&lavc_chmap, &sh_audio->channels) ||
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samplerate != sh_audio->samplerate ||
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sample_format != sh_audio->sample_format) {
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sh_audio->channels = lavc_chmap;
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sh_audio->samplerate = samplerate;
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sh_audio->sample_format = sample_format;
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return 1;
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}
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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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avctx->extradata = av_mallocz(wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
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avctx->extradata_size = wf->cbSize;
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memcpy(avctx->extradata, wf + 1, avctx->extradata_size);
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}
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}
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static int init(sh_audio_t *sh_audio, const char *decoder)
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{
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struct MPOpts *mpopts = sh_audio->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 priv *ctx = talloc_zero(NULL, struct priv);
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sh_audio->context = ctx;
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if (sh_audio->wf && strcmp(decoder, "pcm") == 0) {
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decoder = find_pcm_decoder(tag_map, sh_audio->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_audio->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_tmsg(MSGT_DECAUDIO, MSGL_ERR,
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"Cannot find codec '%s' in libavcodec...\n", decoder);
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uninit(sh_audio);
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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 = avcodec_alloc_frame();
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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_channels = mpopts->audio_output_channels.num;
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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_msg(MSGT_DECVIDEO, MSGL_ERR,
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"ad_lavc: setting AVOptions '%s' failed.\n", opts->avopt);
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uninit(sh_audio);
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return 0;
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}
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}
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lavc_context->codec_tag = sh_audio->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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lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
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FF_INPUT_BUFFER_PADDING_SIZE);
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lavc_context->extradata_size = sh_audio->codecdata_len;
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memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
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lavc_context->extradata_size);
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}
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if (sh_audio->gsh->lav_headers)
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mp_copy_lav_codec_headers(lavc_context, sh_audio->gsh->lav_headers);
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/* open it */
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if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
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mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
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uninit(sh_audio);
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return 0;
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}
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mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
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lavc_codec->name);
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// Decode at least 1 byte: (to get header filled)
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for (int tries = 0;;) {
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int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
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sh_audio->a_buffer_size);
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if (x > 0) {
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sh_audio->a_buffer_len = x;
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break;
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}
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if (++tries >= 5) {
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mp_msg(MSGT_DECAUDIO, MSGL_ERR,
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"ad_lavc: initial decode failed\n");
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uninit(sh_audio);
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return 0;
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}
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}
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sh_audio->i_bps = lavc_context->bit_rate / 8;
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if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
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sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
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int af_sample_fmt =
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af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
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if (af_sample_fmt == AF_FORMAT_UNKNOWN) {
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uninit(sh_audio);
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return 0;
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}
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return 1;
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}
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static void uninit(sh_audio_t *sh)
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{
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struct priv *ctx = sh->context;
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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_tmsg(MSGT_DECVIDEO, MSGL_ERR, "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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avcodec_free_frame(&ctx->avframe);
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talloc_free(ctx);
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sh->context = NULL;
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}
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static int control(sh_audio_t *sh, int cmd, void *arg)
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{
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struct priv *ctx = sh->context;
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switch (cmd) {
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case ADCTRL_RESYNC_STREAM:
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avcodec_flush_buffers(ctx->avctx);
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ctx->output_left = 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 av_always_inline void deplanarize(struct sh_audio *sh)
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{
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struct priv *priv = sh->context;
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uint8_t **planes = priv->avframe->extended_data;
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size_t bps = av_get_bytes_per_sample(priv->avctx->sample_fmt);
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size_t nb_samples = priv->avframe->nb_samples;
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size_t channels = priv->avctx->channels;
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size_t size = bps * nb_samples * channels;
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if (talloc_get_size(priv->output_packed) != size)
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priv->output_packed =
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talloc_realloc_size(priv, priv->output_packed, size);
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reorder_to_packed(priv->output_packed, planes, bps, channels, nb_samples);
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priv->output = priv->output_packed;
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}
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static int decode_new_packet(struct sh_audio *sh)
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{
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struct priv *priv = sh->context;
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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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mpkt = demux_read_packet(sh->gsh);
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if (!mpkt)
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return -1; // error or EOF
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priv->packet = talloc_steal(priv, mpkt);
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int in_len = mpkt->len;
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AVPacket pkt;
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mp_set_av_packet(&pkt, mpkt);
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if (mpkt->pts != MP_NOPTS_VALUE) {
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sh->pts = mpkt->pts;
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sh->pts_bytes = 0;
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}
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int got_frame = 0;
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int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
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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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if (ret < 0) {
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mp_msg(MSGT_DECAUDIO, MSGL_V, "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 0;
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uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
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avctx->channels;
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if (unitsize > 100000)
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abort();
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priv->unitsize = unitsize;
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uint64_t output_left = unitsize * priv->avframe->nb_samples;
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if (output_left > 500000000)
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abort();
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priv->output_left = output_left;
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if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) {
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deplanarize(sh);
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} else {
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priv->output = priv->avframe->data[0];
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}
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mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", in_len,
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priv->output_left);
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return 0;
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}
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static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
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int maxlen)
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{
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struct priv *priv = sh_audio->context;
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AVCodecContext *avctx = priv->avctx;
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int len = -1;
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while (len < minlen) {
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if (!priv->output_left) {
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if (decode_new_packet(sh_audio) < 0)
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break;
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continue;
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}
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if (setup_format(sh_audio, avctx))
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return len;
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int size = (minlen - len + priv->unitsize - 1);
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size -= size % priv->unitsize;
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size = FFMIN(size, priv->output_left);
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if (size > maxlen)
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abort();
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memcpy(buf, priv->output, size);
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priv->output += size;
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priv->output_left -= size;
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if (len < 0)
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len = size;
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else
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len += size;
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buf += size;
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maxlen -= size;
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sh_audio->pts_bytes += size;
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}
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return len;
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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");
|
|
mp_add_decoder(list, "lavc", "mp-pcm", "mp-pcm", "Raw PCM");
|
|
}
|
|
|
|
const struct ad_functions ad_lavc = {
|
|
.name = "lavc",
|
|
.add_decoders = add_decoders,
|
|
.preinit = preinit,
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.control = control,
|
|
.decode_audio = decode_audio,
|
|
};
|