Mercurial > mplayer.hg
annotate libmpcodecs/ad_ffmpeg.c @ 36215:4eb27365d0b0
Add hack to fix compilation issues on OSX with broken libiconv.
Such a hack is not really a good solution, but since the
issue is relatively common it seems worth it for the moment.
author | reimar |
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date | Sun, 09 Jun 2013 13:54:47 +0000 |
parents | 2e8a3822bd84 |
children | 3dfc82c0a678 |
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 */ |
35863 | 18 #define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 |
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19 |
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20 #include <stdio.h> |
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21 #include <stdlib.h> |
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22 #include <unistd.h> |
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23 |
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24 #include "config.h" |
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25 #include "mp_msg.h" |
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26 #include "help_mp.h" |
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27 |
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28 #include "ad_internal.h" |
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29 #include "dec_audio.h" |
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30 #include "av_helpers.h" |
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31 #include "libaf/reorder_ch.h" |
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32 #include "fmt-conversion.h" |
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33 |
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34 static const ad_info_t info = |
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35 { |
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36 "FFmpeg/libavcodec audio decoders", |
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37 "ffmpeg", |
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38 "Nick Kurshev", |
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39 "ffmpeg.sf.net", |
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40 "" |
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41 }; |
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42 |
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43 LIBAD_EXTERN(ffmpeg) |
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44 |
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45 #define assert(x) |
5447 | 46 |
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47 #include "libavcodec/avcodec.h" |
34523 | 48 #include "libavutil/dict.h" |
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49 |
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50 |
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51 static int preinit(sh_audio_t *sh) |
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52 { |
34743 | 53 sh->audio_out_minsize=AF_NCH*AVCODEC_MAX_AUDIO_FRAME_SIZE; |
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54 return 1; |
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55 } |
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56 |
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57 static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context) |
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58 { |
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59 int broken_srate = 0; |
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60 int samplerate = lavc_context->sample_rate; |
35134 | 61 int sample_format = samplefmt2affmt(av_get_packed_sample_fmt(lavc_context->sample_fmt)); |
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62 if (!sample_format) |
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63 sample_format = sh_audio->sample_format; |
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64 if(sh_audio->wf){ |
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65 // If the decoder uses the wrong number of channels all is lost anyway. |
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66 // sh_audio->channels=sh_audio->wf->nChannels; |
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67 |
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68 if (lavc_context->codec_id == AV_CODEC_ID_AAC && |
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69 samplerate == 2*sh_audio->wf->nSamplesPerSec) { |
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70 broken_srate = 1; |
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71 } else if (sh_audio->wf->nSamplesPerSec) |
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72 samplerate=sh_audio->wf->nSamplesPerSec; |
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73 } |
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74 if (lavc_context->channels != sh_audio->channels || |
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75 samplerate != sh_audio->samplerate || |
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76 sample_format != sh_audio->sample_format) { |
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77 sh_audio->channels=lavc_context->channels; |
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78 sh_audio->samplerate=samplerate; |
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79 sh_audio->sample_format = sample_format; |
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80 sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8; |
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81 if (broken_srate) |
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82 mp_msg(MSGT_DECAUDIO, MSGL_WARN, |
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83 "Ignoring broken container sample rate for AAC with SBR\n"); |
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84 return 1; |
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85 } |
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86 return 0; |
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87 } |
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88 |
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89 static int init(sh_audio_t *sh_audio) |
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90 { |
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91 int tries = 0; |
5447 | 92 int x; |
5480 | 93 AVCodecContext *lavc_context; |
94 AVCodec *lavc_codec; | |
34523 | 95 AVDictionary *opts = NULL; |
96 char tmpstr[50]; | |
5447 | 97 |
98 mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); | |
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99 init_avcodec(); |
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100 |
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101 lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll); |
5480 | 102 if(!lavc_codec){ |
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103 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); |
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104 return 0; |
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105 } |
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106 |
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107 lavc_context = avcodec_alloc_context3(lavc_codec); |
5480 | 108 sh_audio->context=lavc_context; |
7940 | 109 |
34563 | 110 snprintf(tmpstr, sizeof(tmpstr), "%f", drc_level); |
34523 | 111 av_dict_set(&opts, "drc_scale", tmpstr, 0); |
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112 lavc_context->sample_rate = sh_audio->samplerate; |
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113 lavc_context->bit_rate = sh_audio->i_bps * 8; |
8590 | 114 if(sh_audio->wf){ |
115 lavc_context->channels = sh_audio->wf->nChannels; | |
116 lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; | |
117 lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; | |
118 lavc_context->block_align = sh_audio->wf->nBlockAlign; | |
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119 lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample; |
8590 | 120 } |
24108 | 121 lavc_context->request_channels = audio_output_channels; |
9547 | 122 lavc_context->codec_tag = sh_audio->format; //FOURCC |
8590 | 123 lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi |
7940 | 124 |
125 /* alloc extra data */ | |
8590 | 126 if (sh_audio->wf && sh_audio->wf->cbSize > 0) { |
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127 lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); |
7940 | 128 lavc_context->extradata_size = sh_audio->wf->cbSize; |
32105 | 129 memcpy(lavc_context->extradata, sh_audio->wf + 1, |
7940 | 130 lavc_context->extradata_size); |
131 } | |
132 | |
14633 | 133 // for QDM2 |
134 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) | |
135 { | |
136 lavc_context->extradata = av_malloc(sh_audio->codecdata_len); | |
137 lavc_context->extradata_size = sh_audio->codecdata_len; | |
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138 memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, |
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139 lavc_context->extradata_size); |
14633 | 140 } |
141 | |
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142 /* open it */ |
34530 | 143 if (avcodec_open2(lavc_context, lavc_codec, &opts) < 0) { |
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144 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); |
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145 return 0; |
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146 } |
34523 | 147 av_dict_free(&opts); |
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148 mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); |
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149 |
8101 | 150 // printf("\nFOURCC: 0x%X\n",sh_audio->format); |
151 if(sh_audio->format==0x3343414D){ | |
152 // MACE 3:1 | |
153 sh_audio->ds->ss_div = 2*3; // 1 samples/packet | |
35257 | 154 sh_audio->ds->ss_mul = sh_audio->wf ? 2*sh_audio->wf->nChannels : 2; // 1 byte*ch/packet |
8101 | 155 } else |
156 if(sh_audio->format==0x3643414D){ | |
157 // MACE 6:1 | |
158 sh_audio->ds->ss_div = 2*6; // 1 samples/packet | |
35257 | 159 sh_audio->ds->ss_mul = sh_audio->wf ? 2*sh_audio->wf->nChannels : 2; // 1 byte*ch/packet |
8101 | 160 } |
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161 |
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162 // Decode at least 1 byte: (to get header filled) |
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163 do { |
29512 | 164 x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); |
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165 } while (x <= 0 && tries++ < 5); |
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166 if(x>0) sh_audio->a_buffer_len=x; |
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167 |
5480 | 168 sh_audio->i_bps=lavc_context->bit_rate/8; |
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169 if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec) |
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170 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; |
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171 |
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172 switch (lavc_context->sample_fmt) { |
35134 | 173 case AV_SAMPLE_FMT_U8: case AV_SAMPLE_FMT_U8P: |
174 case AV_SAMPLE_FMT_S16: case AV_SAMPLE_FMT_S16P: | |
175 case AV_SAMPLE_FMT_S32: case AV_SAMPLE_FMT_S32P: | |
176 case AV_SAMPLE_FMT_FLT: case AV_SAMPLE_FMT_FLTP: | |
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177 break; |
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178 default: |
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179 return 0; |
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180 } |
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181 return 1; |
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182 } |
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183 |
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184 static void uninit(sh_audio_t *sh) |
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185 { |
7940 | 186 AVCodecContext *lavc_context = sh->context; |
187 | |
188 if (avcodec_close(lavc_context) < 0) | |
5447 | 189 mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec); |
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190 av_freep(&lavc_context->extradata); |
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191 av_freep(&lavc_context); |
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192 } |
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193 |
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194 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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195 { |
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196 AVCodecContext *lavc_context = sh->context; |
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197 switch(cmd){ |
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198 case ADCTRL_RESYNC_STREAM: |
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199 avcodec_flush_buffers(lavc_context); |
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200 ds_clear_parser(sh->ds); |
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201 return CONTROL_TRUE; |
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202 } |
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203 return CONTROL_UNKNOWN; |
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204 } |
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205 |
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206 static av_always_inline void copy_samples_planar(size_t bps, |
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207 size_t nb_samples, |
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208 size_t nb_channels, |
35134 | 209 unsigned char *dst, |
210 unsigned char **src) | |
211 { | |
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212 size_t s, c, o = 0; |
35134 | 213 |
214 for (s = 0; s < nb_samples; s++) { | |
215 for (c = 0; c < nb_channels; c++) { | |
216 memcpy(dst, src[c] + o, bps); | |
217 dst += bps; | |
218 } | |
219 o += bps; | |
220 } | |
221 } | |
222 | |
223 static int copy_samples(AVCodecContext *avc, AVFrame *frame, | |
224 unsigned char *buf, int max_size) | |
225 { | |
226 int channels = avc->channels; | |
227 int sample_size = av_get_bytes_per_sample(avc->sample_fmt); | |
228 int size = channels * sample_size * frame->nb_samples; | |
229 | |
230 if (size > max_size) { | |
231 av_log(avc, AV_LOG_ERROR, | |
232 "Buffer overflow while decoding a single frame\n"); | |
233 return AVERROR(EINVAL); /* same as avcodec_decode_audio3 */ | |
234 } | |
235 /* TODO reorder channels at the same time */ | |
236 if (av_sample_fmt_is_planar(avc->sample_fmt)) { | |
237 switch (sample_size) { | |
238 case 1: | |
239 copy_samples_planar(1, frame->nb_samples, channels, | |
240 buf, frame->extended_data); | |
241 break; | |
242 case 2: | |
243 copy_samples_planar(2, frame->nb_samples, channels, | |
244 buf, frame->extended_data); | |
245 break; | |
246 case 4: | |
247 copy_samples_planar(4, frame->nb_samples, channels, | |
248 buf, frame->extended_data); | |
35342 | 249 break; |
35134 | 250 default: |
251 copy_samples_planar(sample_size, frame->nb_samples, channels, | |
252 buf, frame->extended_data); | |
253 } | |
254 } else { | |
255 memcpy(buf, frame->data[0], size); | |
256 } | |
257 return size; | |
258 } | |
259 | |
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260 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) |
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261 { |
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262 unsigned char *start=NULL; |
35134 | 263 int y,len=-1, got_frame; |
264 AVFrame *frame = avcodec_alloc_frame(); | |
265 | |
266 if (!frame) | |
267 return AVERROR(ENOMEM); | |
268 | |
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269 while(len<minlen){ |
29913 | 270 AVPacket pkt; |
23456 | 271 int len2=maxlen; |
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272 double pts; |
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273 int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); |
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274 if(x<=0) { |
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275 start = NULL; |
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276 x = 0; |
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277 ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0); |
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278 if (x <= 0) |
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279 break; // error |
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280 } else { |
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281 int in_size = x; |
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282 int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0); |
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283 sh_audio->ds->buffer_pos -= in_size - consumed; |
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284 } |
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285 |
29913 | 286 av_init_packet(&pkt); |
287 pkt.data = start; | |
288 pkt.size = x; | |
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289 if (pts != MP_NOPTS_VALUE) { |
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290 sh_audio->pts = pts; |
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291 sh_audio->pts_bytes = 0; |
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292 } |
35134 | 293 y=avcodec_decode_audio4(sh_audio->context, frame, &got_frame, &pkt); |
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294 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout); |
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295 // LATM may need many packets to find mux info |
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296 if (y == AVERROR(EAGAIN)) |
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297 continue; |
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298 if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } |
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299 if(!sh_audio->parser && y<x) |
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300 sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) |
35134 | 301 if (!got_frame) |
302 continue; | |
303 len2 = copy_samples(sh_audio->context, frame, buf, maxlen); | |
304 if (len2 < 0) | |
305 return len2; | |
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306 if(len2>0){ |
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307 if (((AVCodecContext *)sh_audio->context)->channels >= 5) { |
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308 int samplesize = av_get_bytes_per_sample(((AVCodecContext *) |
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309 sh_audio->context)->sample_fmt); |
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310 reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, |
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311 AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, |
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312 ((AVCodecContext *)sh_audio->context)->channels, |
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313 len2 / samplesize, samplesize); |
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314 } |
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315 //len=len2;break; |
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316 if(len<0) len=len2; else len+=len2; |
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317 buf+=len2; |
23456 | 318 maxlen -= len2; |
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319 sh_audio->pts_bytes += len2; |
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320 } |
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321 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); |
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322 |
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323 if (setup_format(sh_audio, sh_audio->context)) |
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324 break; |
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325 } |
35134 | 326 |
327 av_free(frame); | |
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328 return len; |
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329 } |