Mercurial > mplayer.hg
view libmpcodecs/ae_lavc.c @ 34757:da38eb1e2069
subassconvert: handle "\r\n" line ends
Previously the code converting text subtitles to ASS format converted newline
characters, and only those, to ASS "new line" markup. If the subtitles
contained "\r\n", the "\r" was thus left in the text. In previous libass
versions the "\r" was not visible, but in the current one it produces an empty
box. Improve the conversion to remove the "\r" in that case. Also treat a lone
"\r" as a newline.
Picked from mplayer2/3e0a2705
author | cboesch |
---|---|
date | Sat, 07 Apr 2012 11:17:09 +0000 |
parents | 505b49b171f4 |
children | 7bf03a973142 |
line wrap: on
line source
/* * This file is part of MPlayer. * * MPlayer is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * MPlayer 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 General Public License for more details. * * You should have received a copy of the GNU General Public License along * with MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include <stdio.h> #include <stdlib.h> #include <inttypes.h> #include <unistd.h> #include <string.h> #include <sys/types.h> #include "config.h" #include "m_option.h" #include "mp_msg.h" #include "libmpdemux/aviheader.h" #include "libmpdemux/ms_hdr.h" #include "stream/stream.h" #include "libmpdemux/muxer.h" #include "ae_lavc.h" #include "av_helpers.h" #include "ve.h" #include "help_mp.h" #include "av_opts.h" #include "libaf/af_format.h" #include "libaf/reorder_ch.h" #include "libavcodec/avcodec.h" #include "libavutil/intreadwrite.h" #include "libavformat/avformat.h" #include "libmpdemux/mp_taglists.h" #include "fmt-conversion.h" static AVCodec *lavc_acodec; static AVCodecContext *lavc_actx; static int compressed_frame_size = 0; static int bind_lavc(audio_encoder_t *encoder, muxer_stream_t *mux_a) { mux_a->wf = malloc(sizeof(WAVEFORMATEX)+lavc_actx->extradata_size+256); mux_a->wf->wFormatTag = lavc_param_atag; mux_a->wf->nChannels = lavc_actx->channels; mux_a->wf->nSamplesPerSec = lavc_actx->sample_rate; mux_a->wf->nAvgBytesPerSec = (lavc_actx->bit_rate / 8); mux_a->avg_rate= lavc_actx->bit_rate; mux_a->h.dwRate = mux_a->wf->nAvgBytesPerSec; if(lavc_actx->block_align) mux_a->h.dwSampleSize = mux_a->h.dwScale = lavc_actx->block_align; else { mux_a->h.dwScale = (mux_a->wf->nAvgBytesPerSec * lavc_actx->frame_size)/ mux_a->wf->nSamplesPerSec; /* for cbr */ if ((mux_a->wf->nAvgBytesPerSec * lavc_actx->frame_size) % mux_a->wf->nSamplesPerSec) { mux_a->h.dwScale = lavc_actx->frame_size; mux_a->h.dwRate = lavc_actx->sample_rate; mux_a->h.dwSampleSize = 0; // Blocksize not constant } else mux_a->h.dwSampleSize = 0; } if(mux_a->h.dwSampleSize) mux_a->wf->nBlockAlign = mux_a->h.dwSampleSize; else mux_a->wf->nBlockAlign = 1; mux_a->h.dwSuggestedBufferSize = (encoder->params.audio_preload*mux_a->wf->nAvgBytesPerSec)/1000; mux_a->h.dwSuggestedBufferSize -= mux_a->h.dwSuggestedBufferSize % mux_a->wf->nBlockAlign; switch(lavc_param_atag) { case 0x11: /* imaadpcm */ mux_a->wf->wBitsPerSample = 4; mux_a->wf->cbSize = 2; AV_WL16(mux_a->wf+1, lavc_actx->frame_size); break; case 0x55: /* mp3 */ mux_a->wf->cbSize = 12; mux_a->wf->wBitsPerSample = 0; /* does not apply */ ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->wID = 1; ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->fdwFlags = 2; ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->nBlockSize = mux_a->wf->nBlockAlign; ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->nFramesPerBlock = 1; ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->nCodecDelay = 0; break; default: mux_a->wf->wBitsPerSample = 0; /* Unknown */ if (lavc_actx->extradata && (lavc_actx->extradata_size > 0)) { memcpy(mux_a->wf+1, lavc_actx->extradata, lavc_actx->extradata_size); mux_a->wf->cbSize = lavc_actx->extradata_size; } else mux_a->wf->cbSize = 0; break; } // Fix allocation mux_a->wf = realloc(mux_a->wf, sizeof(WAVEFORMATEX)+mux_a->wf->cbSize); encoder->min_buffer_size = mux_a->h.dwSuggestedBufferSize; encoder->max_buffer_size = mux_a->h.dwSuggestedBufferSize*2; return 1; } static int encode_lavc(audio_encoder_t *encoder, uint8_t *dest, void *src, int size, int max_size) { int n; if ((encoder->params.channels == 6 || encoder->params.channels == 5) && (!strcmp(lavc_acodec->name,"ac3") || !strcmp(lavc_acodec->name,"libfaac"))) { int isac3 = !strcmp(lavc_acodec->name,"ac3"); int bps = av_get_bytes_per_sample(lavc_actx->sample_fmt); reorder_channel_nch(src, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, isac3 ? AF_CHANNEL_LAYOUT_LAVC_DEFAULT : AF_CHANNEL_LAYOUT_AAC_DEFAULT, encoder->params.channels, size / bps, bps); } n = avcodec_encode_audio(lavc_actx, dest, size, src); compressed_frame_size = n; return n; } static int close_lavc(audio_encoder_t *encoder) { compressed_frame_size = 0; return 1; } static int get_frame_size(audio_encoder_t *encoder) { int sz = compressed_frame_size; compressed_frame_size = 0; return sz; } int mpae_init_lavc(audio_encoder_t *encoder) { encoder->params.samples_per_frame = encoder->params.sample_rate; encoder->params.bitrate = encoder->params.sample_rate * encoder->params.channels * 2 * 8; if(!lavc_param_acodec) { mp_msg(MSGT_MENCODER, MSGL_FATAL, MSGTR_NoLavcAudioCodecName); return 0; } init_avcodec(); lavc_acodec = avcodec_find_encoder_by_name(lavc_param_acodec); if (!lavc_acodec) { mp_msg(MSGT_MENCODER, MSGL_FATAL, MSGTR_LavcAudioCodecNotFound, lavc_param_acodec); return 0; } if(lavc_param_atag == 0) { lavc_param_atag = mp_codec_id2tag(lavc_acodec->id, 0, 1); if(!lavc_param_atag) { mp_msg(MSGT_MENCODER, MSGL_FATAL, "Couldn't find wav tag for specified codec, exit\n"); return 0; } } lavc_actx = avcodec_alloc_context3(lavc_acodec); if(lavc_actx == NULL) { mp_msg(MSGT_MENCODER, MSGL_FATAL, MSGTR_CouldntAllocateLavcContext); return 0; } lavc_actx->codec_id = lavc_acodec->id; // put sample parameters lavc_actx->sample_fmt = AV_SAMPLE_FMT_S16; if (lavc_acodec->sample_fmts) { const enum AVSampleFormat *fmts; lavc_actx->sample_fmt = lavc_acodec->sample_fmts[0]; // fallback to first format for (fmts = lavc_acodec->sample_fmts; *fmts != AV_SAMPLE_FMT_NONE; fmts++) { if (samplefmt2affmt(*fmts) == encoder->params.sample_format) { // preferred format found lavc_actx->sample_fmt = *fmts; break; } } } encoder->input_format = samplefmt2affmt(lavc_actx->sample_fmt); lavc_actx->channels = encoder->params.channels; lavc_actx->sample_rate = encoder->params.sample_rate; lavc_actx->time_base.num = 1; lavc_actx->time_base.den = encoder->params.sample_rate; if(lavc_param_abitrate<1000) lavc_actx->bit_rate = encoder->params.bitrate = lavc_param_abitrate * 1000; else lavc_actx->bit_rate = encoder->params.bitrate = lavc_param_abitrate; if(lavc_param_audio_avopt){ if(parse_avopts(lavc_actx, lavc_param_audio_avopt) < 0){ mp_msg(MSGT_MENCODER,MSGL_ERR, "Your options /%s/ look like gibberish to me pal\n", lavc_param_audio_avopt); return 0; } } /* * Special case for adpcm_ima_wav. * The bitrate is only dependent on samplerate. * We have to known frame_size and block_align in advance, * so I just copied the code from libavcodec/adpcm.c * * However, ms adpcm_ima_wav uses a block_align of 2048, * lavc defaults to 1024 */ if(lavc_param_atag == 0x11) { int blkalign = 2048; int framesize = (blkalign - 4 * lavc_actx->channels) * 8 / (4 * lavc_actx->channels) + 1; lavc_actx->bit_rate = lavc_actx->sample_rate*8*blkalign/framesize; } if((lavc_param_audio_global_header&1) /*|| (video_global_header==0 && (oc->oformat->flags & AVFMT_GLOBALHEADER))*/){ lavc_actx->flags |= CODEC_FLAG_GLOBAL_HEADER; } if(lavc_param_audio_global_header&2){ lavc_actx->flags2 |= CODEC_FLAG2_LOCAL_HEADER; } if(avcodec_open2(lavc_actx, lavc_acodec, NULL) < 0) { mp_msg(MSGT_MENCODER, MSGL_FATAL, MSGTR_CouldntOpenCodec, lavc_param_acodec, lavc_param_abitrate); return 0; } if(lavc_param_atag == 0x11) { lavc_actx->block_align = 2048; lavc_actx->frame_size = (lavc_actx->block_align - 4 * lavc_actx->channels) * 8 / (4 * lavc_actx->channels) + 1; } encoder->decode_buffer_size = lavc_actx->frame_size * av_get_bytes_per_sample(lavc_actx->sample_fmt) * encoder->params.channels; while (encoder->decode_buffer_size < 1024) encoder->decode_buffer_size *= 2; encoder->bind = bind_lavc; encoder->get_frame_size = get_frame_size; encoder->encode = encode_lavc; encoder->close = close_lavc; return 1; }