Mercurial > mplayer.hg
annotate libmpcodecs/ad_ffmpeg.c @ 33672:e576232a39d5
Prevent balance from hopping.
Only recalculate the balance if the balance has changed, not if just
the volume has changed.
Because (at least with my soundcard) not all volume values can be
stored, but seem to be mapped onto a discrete value set, recalculation
the balance from the volume isn't accurate enough.
author | ib |
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date | Tue, 28 Jun 2011 18:16:06 +0000 |
parents | e2382d88fb7f |
children | 30f5e5cd3676 |
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 "vd_ffmpeg.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" |
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47 |
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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 setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context) |
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56 { |
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57 int broken_srate = 0; |
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58 int samplerate = lavc_context->sample_rate; |
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59 int sample_format = samplefmt2affmt(lavc_context->sample_fmt); |
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60 if (!sample_format) |
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61 sample_format = sh_audio->sample_format; |
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62 if(sh_audio->wf){ |
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63 // If the decoder uses the wrong number of channels all is lost anyway. |
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64 // sh_audio->channels=sh_audio->wf->nChannels; |
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65 |
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66 if (lavc_context->codec_id == CODEC_ID_AAC && |
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67 samplerate == 2*sh_audio->wf->nSamplesPerSec) { |
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68 broken_srate = 1; |
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69 } else if (sh_audio->wf->nSamplesPerSec) |
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70 samplerate=sh_audio->wf->nSamplesPerSec; |
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71 } |
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72 if (lavc_context->channels != sh_audio->channels || |
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73 samplerate != sh_audio->samplerate || |
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74 sample_format != sh_audio->sample_format) { |
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75 sh_audio->channels=lavc_context->channels; |
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76 sh_audio->samplerate=samplerate; |
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77 sh_audio->sample_format = sample_format; |
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78 sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8; |
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79 if (broken_srate) |
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80 mp_msg(MSGT_DECAUDIO, MSGL_WARN, |
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81 "Ignoring broken container sample rate for AAC with SBR\n"); |
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82 return 1; |
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83 } |
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84 return 0; |
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85 } |
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86 |
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87 static int init(sh_audio_t *sh_audio) |
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88 { |
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89 int tries = 0; |
5447 | 90 int x; |
5480 | 91 AVCodecContext *lavc_context; |
92 AVCodec *lavc_codec; | |
5447 | 93 |
94 mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); | |
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95 init_avcodec(); |
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96 |
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97 lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll); |
5480 | 98 if(!lavc_codec){ |
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99 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); |
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100 return 0; |
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101 } |
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102 |
7940 | 103 lavc_context = avcodec_alloc_context(); |
5480 | 104 sh_audio->context=lavc_context; |
7940 | 105 |
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106 lavc_context->drc_scale = drc_level; |
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107 lavc_context->sample_rate = sh_audio->samplerate; |
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108 lavc_context->bit_rate = sh_audio->i_bps * 8; |
8590 | 109 if(sh_audio->wf){ |
110 lavc_context->channels = sh_audio->wf->nChannels; | |
111 lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; | |
112 lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; | |
113 lavc_context->block_align = sh_audio->wf->nBlockAlign; | |
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114 lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample; |
8590 | 115 } |
24108 | 116 lavc_context->request_channels = audio_output_channels; |
9547 | 117 lavc_context->codec_tag = sh_audio->format; //FOURCC |
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118 lavc_context->codec_type = AVMEDIA_TYPE_AUDIO; |
8590 | 119 lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi |
7940 | 120 |
121 /* alloc extra data */ | |
8590 | 122 if (sh_audio->wf && sh_audio->wf->cbSize > 0) { |
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123 lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); |
7940 | 124 lavc_context->extradata_size = sh_audio->wf->cbSize; |
32105 | 125 memcpy(lavc_context->extradata, sh_audio->wf + 1, |
7940 | 126 lavc_context->extradata_size); |
127 } | |
128 | |
14633 | 129 // for QDM2 |
130 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) | |
131 { | |
132 lavc_context->extradata = av_malloc(sh_audio->codecdata_len); | |
133 lavc_context->extradata_size = sh_audio->codecdata_len; | |
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134 memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, |
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135 lavc_context->extradata_size); |
14633 | 136 } |
137 | |
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138 /* open it */ |
5480 | 139 if (avcodec_open(lavc_context, lavc_codec) < 0) { |
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140 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); |
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141 return 0; |
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142 } |
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143 mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); |
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144 |
8101 | 145 // printf("\nFOURCC: 0x%X\n",sh_audio->format); |
146 if(sh_audio->format==0x3343414D){ | |
147 // MACE 3:1 | |
148 sh_audio->ds->ss_div = 2*3; // 1 samples/packet | |
8108 | 149 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet |
8101 | 150 } else |
151 if(sh_audio->format==0x3643414D){ | |
152 // MACE 6:1 | |
153 sh_audio->ds->ss_div = 2*6; // 1 samples/packet | |
8108 | 154 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet |
8101 | 155 } |
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156 |
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157 // Decode at least 1 byte: (to get header filled) |
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158 do { |
29512 | 159 x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); |
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160 } while (x <= 0 && tries++ < 5); |
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161 if(x>0) sh_audio->a_buffer_len=x; |
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162 |
5480 | 163 sh_audio->i_bps=lavc_context->bit_rate/8; |
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164 if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec) |
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165 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; |
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166 |
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167 switch (lavc_context->sample_fmt) { |
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168 case AV_SAMPLE_FMT_U8: |
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169 case AV_SAMPLE_FMT_S16: |
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170 case AV_SAMPLE_FMT_S32: |
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171 case AV_SAMPLE_FMT_FLT: |
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172 break; |
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173 default: |
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174 return 0; |
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175 } |
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176 return 1; |
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177 } |
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178 |
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179 static void uninit(sh_audio_t *sh) |
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180 { |
7940 | 181 AVCodecContext *lavc_context = sh->context; |
182 | |
183 if (avcodec_close(lavc_context) < 0) | |
5447 | 184 mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec); |
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185 av_freep(&lavc_context->extradata); |
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186 av_freep(&lavc_context); |
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187 } |
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188 |
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189 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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190 { |
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191 AVCodecContext *lavc_context = sh->context; |
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192 switch(cmd){ |
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193 case ADCTRL_RESYNC_STREAM: |
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194 avcodec_flush_buffers(lavc_context); |
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195 ds_clear_parser(sh->ds); |
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196 return CONTROL_TRUE; |
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197 } |
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198 return CONTROL_UNKNOWN; |
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199 } |
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200 |
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201 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) |
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202 { |
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203 unsigned char *start=NULL; |
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204 int y,len=-1; |
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205 while(len<minlen){ |
29913 | 206 AVPacket pkt; |
23456 | 207 int len2=maxlen; |
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208 double pts; |
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209 int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); |
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210 if(x<=0) { |
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211 start = NULL; |
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212 x = 0; |
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213 ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0); |
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214 if (x <= 0) |
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215 break; // error |
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216 } else { |
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217 int in_size = x; |
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218 int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0); |
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219 sh_audio->ds->buffer_pos -= in_size - consumed; |
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220 } |
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221 |
29913 | 222 av_init_packet(&pkt); |
223 pkt.data = start; | |
224 pkt.size = x; | |
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225 if (pts != MP_NOPTS_VALUE) { |
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226 sh_audio->pts = pts; |
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227 sh_audio->pts_bytes = 0; |
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228 } |
29913 | 229 y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt); |
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230 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout); |
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231 // LATM may need many packets to find mux info |
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232 if (y == AVERROR(EAGAIN)) |
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233 continue; |
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234 if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } |
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235 if(!sh_audio->parser && y<x) |
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236 sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) |
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237 if(len2>0){ |
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238 if (((AVCodecContext *)sh_audio->context)->channels >= 5) { |
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239 int samplesize = av_get_bits_per_sample_fmt(((AVCodecContext *) |
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240 sh_audio->context)->sample_fmt) / 8; |
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241 reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, |
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242 AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, |
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243 ((AVCodecContext *)sh_audio->context)->channels, |
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244 len2 / samplesize, samplesize); |
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245 } |
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246 //len=len2;break; |
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247 if(len<0) len=len2; else len+=len2; |
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248 buf+=len2; |
23456 | 249 maxlen -= len2; |
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250 sh_audio->pts_bytes += len2; |
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251 } |
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252 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); |
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253 |
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254 if (setup_format(sh_audio, sh_audio->context)) |
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255 break; |
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256 } |
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257 return len; |
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258 } |