Mercurial > mplayer.hg
annotate libmpcodecs/ad_ffmpeg.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 |
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date | Sat, 07 Apr 2012 11:17:09 +0000 |
parents | e48eefbb92d2 |
children | 2b037cee0795 |
rev | line source |
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1 /* |
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2 * This file is part of MPlayer. |
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3 * |
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4 * MPlayer is free software; you can redistribute it and/or modify |
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5 * it under the terms of the GNU General Public License as published by |
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6 * the Free Software Foundation; either version 2 of the License, or |
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7 * (at your option) any later version. |
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8 * |
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9 * MPlayer is distributed in the hope that it will be useful, |
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10 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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12 * GNU General Public License for more details. |
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13 * |
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14 * You should have received a copy of the GNU General Public License along |
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15 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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16 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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17 */ |
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18 |
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19 #include <stdio.h> |
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20 #include <stdlib.h> |
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21 #include <unistd.h> |
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22 |
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23 #include "config.h" |
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24 #include "mp_msg.h" |
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25 #include "help_mp.h" |
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26 |
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27 #include "ad_internal.h" |
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28 #include "dec_audio.h" |
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29 #include "av_helpers.h" |
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30 #include "libaf/reorder_ch.h" |
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31 #include "fmt-conversion.h" |
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32 |
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33 static const ad_info_t info = |
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34 { |
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35 "FFmpeg/libavcodec audio decoders", |
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36 "ffmpeg", |
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37 "Nick Kurshev", |
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38 "ffmpeg.sf.net", |
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39 "" |
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40 }; |
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41 |
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42 LIBAD_EXTERN(ffmpeg) |
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43 |
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44 #define assert(x) |
5447 | 45 |
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46 #include "libavcodec/avcodec.h" |
34523 | 47 #include "libavutil/dict.h" |
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48 |
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49 |
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50 static int preinit(sh_audio_t *sh) |
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51 { |
34743 | 52 sh->audio_out_minsize=AF_NCH*AVCODEC_MAX_AUDIO_FRAME_SIZE; |
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53 return 1; |
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54 } |
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55 |
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56 static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context) |
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57 { |
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58 int broken_srate = 0; |
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59 int samplerate = lavc_context->sample_rate; |
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60 int sample_format = samplefmt2affmt(lavc_context->sample_fmt); |
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61 if (!sample_format) |
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62 sample_format = sh_audio->sample_format; |
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63 if(sh_audio->wf){ |
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64 // If the decoder uses the wrong number of channels all is lost anyway. |
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65 // sh_audio->channels=sh_audio->wf->nChannels; |
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66 |
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67 if (lavc_context->codec_id == CODEC_ID_AAC && |
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68 samplerate == 2*sh_audio->wf->nSamplesPerSec) { |
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69 broken_srate = 1; |
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70 } else if (sh_audio->wf->nSamplesPerSec) |
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71 samplerate=sh_audio->wf->nSamplesPerSec; |
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72 } |
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73 if (lavc_context->channels != sh_audio->channels || |
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74 samplerate != sh_audio->samplerate || |
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75 sample_format != sh_audio->sample_format) { |
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76 sh_audio->channels=lavc_context->channels; |
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77 sh_audio->samplerate=samplerate; |
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78 sh_audio->sample_format = sample_format; |
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79 sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8; |
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80 if (broken_srate) |
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81 mp_msg(MSGT_DECAUDIO, MSGL_WARN, |
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82 "Ignoring broken container sample rate for AAC with SBR\n"); |
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83 return 1; |
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84 } |
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85 return 0; |
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86 } |
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87 |
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88 static int init(sh_audio_t *sh_audio) |
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89 { |
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90 int tries = 0; |
5447 | 91 int x; |
5480 | 92 AVCodecContext *lavc_context; |
93 AVCodec *lavc_codec; | |
34523 | 94 AVDictionary *opts = NULL; |
95 char tmpstr[50]; | |
5447 | 96 |
97 mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); | |
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98 init_avcodec(); |
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99 |
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100 lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll); |
5480 | 101 if(!lavc_codec){ |
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102 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); |
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103 return 0; |
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104 } |
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105 |
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106 lavc_context = avcodec_alloc_context3(lavc_codec); |
5480 | 107 sh_audio->context=lavc_context; |
7940 | 108 |
34563 | 109 snprintf(tmpstr, sizeof(tmpstr), "%f", drc_level); |
34523 | 110 av_dict_set(&opts, "drc_scale", tmpstr, 0); |
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111 lavc_context->sample_rate = sh_audio->samplerate; |
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112 lavc_context->bit_rate = sh_audio->i_bps * 8; |
8590 | 113 if(sh_audio->wf){ |
114 lavc_context->channels = sh_audio->wf->nChannels; | |
115 lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; | |
116 lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; | |
117 lavc_context->block_align = sh_audio->wf->nBlockAlign; | |
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118 lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample; |
8590 | 119 } |
24108 | 120 lavc_context->request_channels = audio_output_channels; |
9547 | 121 lavc_context->codec_tag = sh_audio->format; //FOURCC |
8590 | 122 lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi |
7940 | 123 |
124 /* alloc extra data */ | |
8590 | 125 if (sh_audio->wf && sh_audio->wf->cbSize > 0) { |
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126 lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); |
7940 | 127 lavc_context->extradata_size = sh_audio->wf->cbSize; |
32105 | 128 memcpy(lavc_context->extradata, sh_audio->wf + 1, |
7940 | 129 lavc_context->extradata_size); |
130 } | |
131 | |
14633 | 132 // for QDM2 |
133 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) | |
134 { | |
135 lavc_context->extradata = av_malloc(sh_audio->codecdata_len); | |
136 lavc_context->extradata_size = sh_audio->codecdata_len; | |
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137 memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, |
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138 lavc_context->extradata_size); |
14633 | 139 } |
140 | |
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141 /* open it */ |
34530 | 142 if (avcodec_open2(lavc_context, lavc_codec, &opts) < 0) { |
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143 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); |
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144 return 0; |
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145 } |
34523 | 146 av_dict_free(&opts); |
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147 mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); |
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148 |
8101 | 149 // printf("\nFOURCC: 0x%X\n",sh_audio->format); |
150 if(sh_audio->format==0x3343414D){ | |
151 // MACE 3:1 | |
152 sh_audio->ds->ss_div = 2*3; // 1 samples/packet | |
8108 | 153 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet |
8101 | 154 } else |
155 if(sh_audio->format==0x3643414D){ | |
156 // MACE 6:1 | |
157 sh_audio->ds->ss_div = 2*6; // 1 samples/packet | |
8108 | 158 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet |
8101 | 159 } |
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160 |
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161 // Decode at least 1 byte: (to get header filled) |
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162 do { |
29512 | 163 x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); |
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164 } while (x <= 0 && tries++ < 5); |
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165 if(x>0) sh_audio->a_buffer_len=x; |
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166 |
5480 | 167 sh_audio->i_bps=lavc_context->bit_rate/8; |
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168 if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec) |
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169 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; |
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170 |
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171 switch (lavc_context->sample_fmt) { |
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172 case AV_SAMPLE_FMT_U8: |
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173 case AV_SAMPLE_FMT_S16: |
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174 case AV_SAMPLE_FMT_S32: |
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175 case AV_SAMPLE_FMT_FLT: |
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176 break; |
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177 default: |
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178 return 0; |
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179 } |
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180 return 1; |
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181 } |
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182 |
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183 static void uninit(sh_audio_t *sh) |
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184 { |
7940 | 185 AVCodecContext *lavc_context = sh->context; |
186 | |
187 if (avcodec_close(lavc_context) < 0) | |
5447 | 188 mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec); |
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189 av_freep(&lavc_context->extradata); |
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190 av_freep(&lavc_context); |
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191 } |
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192 |
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193 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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194 { |
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195 AVCodecContext *lavc_context = sh->context; |
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196 switch(cmd){ |
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197 case ADCTRL_RESYNC_STREAM: |
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198 avcodec_flush_buffers(lavc_context); |
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199 ds_clear_parser(sh->ds); |
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200 return CONTROL_TRUE; |
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201 } |
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202 return CONTROL_UNKNOWN; |
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203 } |
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204 |
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205 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) |
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206 { |
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207 unsigned char *start=NULL; |
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208 int y,len=-1; |
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209 while(len<minlen){ |
29913 | 210 AVPacket pkt; |
23456 | 211 int len2=maxlen; |
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212 double pts; |
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213 int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); |
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214 if(x<=0) { |
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215 start = NULL; |
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216 x = 0; |
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217 ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0); |
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218 if (x <= 0) |
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219 break; // error |
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220 } else { |
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221 int in_size = x; |
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222 int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0); |
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223 sh_audio->ds->buffer_pos -= in_size - consumed; |
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224 } |
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225 |
29913 | 226 av_init_packet(&pkt); |
227 pkt.data = start; | |
228 pkt.size = x; | |
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229 if (pts != MP_NOPTS_VALUE) { |
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230 sh_audio->pts = pts; |
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231 sh_audio->pts_bytes = 0; |
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232 } |
29913 | 233 y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt); |
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234 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout); |
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235 // LATM may need many packets to find mux info |
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236 if (y == AVERROR(EAGAIN)) |
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237 continue; |
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238 if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } |
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239 if(!sh_audio->parser && y<x) |
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240 sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) |
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241 if(len2>0){ |
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242 if (((AVCodecContext *)sh_audio->context)->channels >= 5) { |
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243 int samplesize = av_get_bytes_per_sample(((AVCodecContext *) |
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244 sh_audio->context)->sample_fmt); |
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245 reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, |
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246 AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, |
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247 ((AVCodecContext *)sh_audio->context)->channels, |
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248 len2 / samplesize, samplesize); |
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249 } |
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250 //len=len2;break; |
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251 if(len<0) len=len2; else len+=len2; |
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252 buf+=len2; |
23456 | 253 maxlen -= len2; |
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254 sh_audio->pts_bytes += len2; |
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255 } |
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256 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); |
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257 |
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258 if (setup_format(sh_audio, sh_audio->context)) |
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259 break; |
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260 } |
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261 return len; |
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262 } |