Mercurial > mplayer.hg
annotate libmpcodecs/ad_ffmpeg.c @ 31215:d0f70692a140
Make 3dnow and MMX code compile for 64 bit (just stupid search-and-replace).
They are not compiled/enabled on 64 bit though, since they are vastly slower
but would still be selected by default.
author | reimar |
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date | Sun, 30 May 2010 10:01:40 +0000 |
parents | b7040d298579 |
children | 002cdc635c03 |
rev | line source |
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30421
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Add missing license headers to all files in the libmpcodecs directory.
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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" |
25315
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Fix all current known multi-channel wrong order problems by adding
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28 #include "libaf/reorder_ch.h" |
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29 |
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Hack around libavutil/bswap.h compilation problems due to always_inline undefined.
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30 #include "mpbswap.h" |
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31 |
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32 static const ad_info_t info = |
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33 { |
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34 "FFmpeg/libavcodec audio decoders", |
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35 "ffmpeg", |
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36 "Nick Kurshev", |
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37 "ffmpeg.sf.net", |
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38 "" |
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39 }; |
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40 |
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41 LIBAD_EXTERN(ffmpeg) |
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42 |
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43 #define assert(x) |
5447 | 44 |
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45 #include "libavcodec/avcodec.h" |
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46 |
25962 | 47 extern int avcodec_initialized; |
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48 |
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49 static int preinit(sh_audio_t *sh) |
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50 { |
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51 sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE; |
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52 return 1; |
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53 } |
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54 |
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55 static int init(sh_audio_t *sh_audio) |
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56 { |
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57 int tries = 0; |
5447 | 58 int x; |
5480 | 59 AVCodecContext *lavc_context; |
60 AVCodec *lavc_codec; | |
5447 | 61 |
62 mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); | |
25962 | 63 if(!avcodec_initialized){ |
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64 avcodec_init(); |
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65 avcodec_register_all(); |
25962 | 66 avcodec_initialized=1; |
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67 } |
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68 |
5480 | 69 lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll); |
70 if(!lavc_codec){ | |
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71 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); |
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72 return 0; |
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73 } |
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74 |
7940 | 75 lavc_context = avcodec_alloc_context(); |
5480 | 76 sh_audio->context=lavc_context; |
7940 | 77 |
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78 lavc_context->sample_rate = sh_audio->samplerate; |
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79 lavc_context->bit_rate = sh_audio->i_bps * 8; |
8590 | 80 if(sh_audio->wf){ |
81 lavc_context->channels = sh_audio->wf->nChannels; | |
82 lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; | |
83 lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; | |
84 lavc_context->block_align = sh_audio->wf->nBlockAlign; | |
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85 lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample; |
8590 | 86 } |
24108 | 87 lavc_context->request_channels = audio_output_channels; |
9547 | 88 lavc_context->codec_tag = sh_audio->format; //FOURCC |
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89 lavc_context->codec_type = CODEC_TYPE_AUDIO; |
8590 | 90 lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi |
7940 | 91 |
92 /* alloc extra data */ | |
8590 | 93 if (sh_audio->wf && sh_audio->wf->cbSize > 0) { |
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94 lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); |
7940 | 95 lavc_context->extradata_size = sh_audio->wf->cbSize; |
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96 memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX), |
7940 | 97 lavc_context->extradata_size); |
98 } | |
99 | |
14633 | 100 // for QDM2 |
101 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) | |
102 { | |
103 lavc_context->extradata = av_malloc(sh_audio->codecdata_len); | |
104 lavc_context->extradata_size = sh_audio->codecdata_len; | |
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105 memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, |
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106 lavc_context->extradata_size); |
14633 | 107 } |
108 | |
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109 /* open it */ |
5480 | 110 if (avcodec_open(lavc_context, lavc_codec) < 0) { |
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111 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); |
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112 return 0; |
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113 } |
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114 mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); |
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115 |
8101 | 116 // printf("\nFOURCC: 0x%X\n",sh_audio->format); |
117 if(sh_audio->format==0x3343414D){ | |
118 // MACE 3:1 | |
119 sh_audio->ds->ss_div = 2*3; // 1 samples/packet | |
8108 | 120 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet |
8101 | 121 } else |
122 if(sh_audio->format==0x3643414D){ | |
123 // MACE 6:1 | |
124 sh_audio->ds->ss_div = 2*6; // 1 samples/packet | |
8108 | 125 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet |
8101 | 126 } |
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127 |
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128 // Decode at least 1 byte: (to get header filled) |
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129 do { |
29512 | 130 x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); |
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131 } while (x <= 0 && tries++ < 5); |
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132 if(x>0) sh_audio->a_buffer_len=x; |
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133 |
5480 | 134 sh_audio->channels=lavc_context->channels; |
135 sh_audio->samplerate=lavc_context->sample_rate; | |
136 sh_audio->i_bps=lavc_context->bit_rate/8; | |
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137 switch (lavc_context->sample_fmt) { |
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138 case SAMPLE_FMT_U8: sh_audio->sample_format = AF_FORMAT_U8; break; |
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139 case SAMPLE_FMT_S16: sh_audio->sample_format = AF_FORMAT_S16_NE; break; |
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140 case SAMPLE_FMT_S32: sh_audio->sample_format = AF_FORMAT_S32_NE; break; |
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141 case SAMPLE_FMT_FLT: sh_audio->sample_format = AF_FORMAT_FLOAT_NE; break; |
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142 default: |
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143 mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n"); |
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144 return 0; |
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145 } |
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146 if(sh_audio->wf){ |
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147 // If the decoder uses the wrong number of channels all is lost anyway. |
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148 // sh_audio->channels=sh_audio->wf->nChannels; |
31010 | 149 |
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150 if (lavc_context->codec_id == CODEC_ID_AAC && |
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151 sh_audio->samplerate == 2*sh_audio->wf->nSamplesPerSec) { |
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152 mp_msg(MSGT_DECAUDIO, MSGL_WARN, |
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153 "Ignoring broken container sample rate for ACC with SBR\n"); |
31010 | 154 } else if (sh_audio->wf->nSamplesPerSec) |
155 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; | |
156 | |
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157 if (sh_audio->wf->nAvgBytesPerSec) |
31010 | 158 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; |
17995
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Trust the demuxer instead of the decoder for samplerate detection.
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159 } |
28955
ee06f3a8b0d5
Support FFmpeg codecs that decode to other formats than S16.
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160 sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8; |
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161 return 1; |
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162 } |
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163 |
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164 static void uninit(sh_audio_t *sh) |
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165 { |
7940 | 166 AVCodecContext *lavc_context = sh->context; |
167 | |
168 if (avcodec_close(lavc_context) < 0) | |
5447 | 169 mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec); |
14431
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170 av_freep(&lavc_context->extradata); |
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171 av_freep(&lavc_context); |
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172 } |
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173 |
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174 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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175 { |
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176 AVCodecContext *lavc_context = sh->context; |
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177 switch(cmd){ |
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178 case ADCTRL_RESYNC_STREAM: |
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179 avcodec_flush_buffers(lavc_context); |
30379
0140d505dd91
Reset the parser on seek. Should fix some cases of audio "blips" after seeking.
reimar
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180 ds_clear_parser(sh->ds); |
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181 return CONTROL_TRUE; |
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182 } |
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183 return CONTROL_UNKNOWN; |
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184 } |
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185 |
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186 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) |
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187 { |
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188 unsigned char *start=NULL; |
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189 int y,len=-1; |
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190 while(len<minlen){ |
29913 | 191 AVPacket pkt; |
23456 | 192 int len2=maxlen; |
18242
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rtognimp
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193 double pts; |
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194 int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); |
30088
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195 if(x<=0) { |
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196 start = NULL; |
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197 x = 0; |
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198 ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0); |
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199 if (x <= 0) |
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200 break; // error |
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reimar
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201 } else { |
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reimar
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202 int in_size = x; |
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203 int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0); |
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204 sh_audio->ds->buffer_pos -= in_size - consumed; |
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205 } |
29913 | 206 av_init_packet(&pkt); |
207 pkt.data = start; | |
208 pkt.size = x; | |
18242
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209 if (pts != MP_NOPTS_VALUE) { |
caac2ca98168
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210 sh_audio->pts = pts; |
caac2ca98168
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rtognimp
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211 sh_audio->pts_bytes = 0; |
caac2ca98168
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212 } |
29913 | 213 y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt); |
11977
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214 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout); |
5340
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215 if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } |
30272
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We only need to disable seeking back in ad_ffmpeg when we actually _use_
reimar
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216 if(!sh_audio->parser && y<x) |
30088
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217 sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) |
5340
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218 if(len2>0){ |
25315
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Fix all current known multi-channel wrong order problems by adding
ulion
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219 if (((AVCodecContext *)sh_audio->context)->channels >= 5) { |
29491
99eda963d27a
Fix incorrect channel ordering for lavc audio codecs (specifically ffac3,
tack
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220 int samplesize = av_get_bits_per_sample_format(((AVCodecContext *) |
99eda963d27a
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tack
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221 sh_audio->context)->sample_fmt) / 8; |
29779
282ea4fbe87d
ad_ffmpeg: Fix channel layout for ffvorbis and ffaac
tack
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222 reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, |
25315
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ulion
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223 AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, |
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ulion
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224 ((AVCodecContext *)sh_audio->context)->channels, |
29491
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Fix incorrect channel ordering for lavc audio codecs (specifically ffac3,
tack
parents:
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225 len2 / samplesize, samplesize); |
25315
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ulion
parents:
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226 } |
5340
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227 //len=len2;break; |
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228 if(len<0) len=len2; else len+=len2; |
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229 buf+=len2; |
23456 | 230 maxlen -= len2; |
18242
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rtognimp
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231 sh_audio->pts_bytes += len2; |
5340
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232 } |
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233 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); |
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234 } |
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235 return len; |
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236 } |