Mercurial > mplayer.hg
annotate libmpcodecs/ad_ffmpeg.c @ 37110:4e00eb3f4c76
ad_ffmpeg: Set up audio format already during init.
It seems otherwise mencoder can get confused.
author | reimar |
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date | Tue, 20 May 2014 19:29:31 +0000 |
parents | a706bf2ed700 |
children |
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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-afm/-vfm help implemenetd, some cosmetics of ad/vd codec names/comments
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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 |
36369 | 50 struct adctx { |
51 int last_samplerate; | |
52 int srate_changed; | |
53 }; | |
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54 |
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55 static int preinit(sh_audio_t *sh) |
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56 { |
34743 | 57 sh->audio_out_minsize=AF_NCH*AVCODEC_MAX_AUDIO_FRAME_SIZE; |
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58 return 1; |
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59 } |
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60 |
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61 static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context) |
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62 { |
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63 int broken_srate = 0; |
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64 int samplerate = lavc_context->sample_rate; |
35134 | 65 int sample_format = samplefmt2affmt(av_get_packed_sample_fmt(lavc_context->sample_fmt)); |
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66 if (!sample_format) |
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67 sample_format = sh_audio->sample_format; |
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68 if(sh_audio->wf){ |
36369 | 69 struct adctx *c = lavc_context->opaque; |
70 c->srate_changed |= c->last_samplerate && c->last_samplerate != samplerate; | |
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71 c->last_samplerate = samplerate; |
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72 // If the decoder uses the wrong number of channels all is lost anyway. |
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73 // sh_audio->channels=sh_audio->wf->nChannels; |
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74 |
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75 if (lavc_context->codec_id == AV_CODEC_ID_AAC && |
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76 samplerate == 2*sh_audio->wf->nSamplesPerSec) { |
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77 broken_srate = 1; |
36369 | 78 } else if (sh_audio->wf->nSamplesPerSec && !c->srate_changed) |
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79 samplerate=sh_audio->wf->nSamplesPerSec; |
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80 } |
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81 if (lavc_context->channels != sh_audio->channels || |
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82 samplerate != sh_audio->samplerate || |
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83 sample_format != sh_audio->sample_format) { |
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84 sh_audio->channels=lavc_context->channels; |
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85 sh_audio->samplerate=samplerate; |
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86 sh_audio->sample_format = sample_format; |
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87 sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8; |
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88 if (broken_srate) |
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89 mp_msg(MSGT_DECAUDIO, MSGL_WARN, |
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90 "Ignoring broken container sample rate for AAC with SBR\n"); |
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91 return 1; |
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92 } |
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93 return 0; |
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94 } |
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95 |
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96 static int init(sh_audio_t *sh_audio) |
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97 { |
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98 int tries = 0; |
5447 | 99 int x; |
5480 | 100 AVCodecContext *lavc_context; |
101 AVCodec *lavc_codec; | |
34523 | 102 AVDictionary *opts = NULL; |
103 char tmpstr[50]; | |
5447 | 104 |
105 mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); | |
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106 init_avcodec(); |
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107 |
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108 lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll); |
5480 | 109 if(!lavc_codec){ |
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110 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); |
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111 return 0; |
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112 } |
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113 |
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114 lavc_context = avcodec_alloc_context3(lavc_codec); |
5480 | 115 sh_audio->context=lavc_context; |
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116 lavc_context->opaque = av_mallocz(sizeof(struct adctx)); |
7940 | 117 |
34563 | 118 snprintf(tmpstr, sizeof(tmpstr), "%f", drc_level); |
34523 | 119 av_dict_set(&opts, "drc_scale", tmpstr, 0); |
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120 lavc_context->sample_rate = sh_audio->samplerate; |
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121 lavc_context->bit_rate = sh_audio->i_bps * 8; |
8590 | 122 if(sh_audio->wf){ |
123 lavc_context->channels = sh_audio->wf->nChannels; | |
124 lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; | |
125 lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; | |
126 lavc_context->block_align = sh_audio->wf->nBlockAlign; | |
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127 lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample; |
8590 | 128 } |
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129 lavc_context->channel_layout = sh_audio->channel_layout; |
36770 | 130 if (audio_output_channels == 1) |
131 lavc_context->request_channel_layout = AV_CH_LAYOUT_MONO; | |
132 else if (audio_output_channels == 2) | |
133 lavc_context->request_channel_layout = AV_CH_LAYOUT_STEREO; | |
9547 | 134 lavc_context->codec_tag = sh_audio->format; //FOURCC |
8590 | 135 lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi |
7940 | 136 |
137 /* alloc extra data */ | |
8590 | 138 if (sh_audio->wf && sh_audio->wf->cbSize > 0) { |
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139 lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); |
7940 | 140 lavc_context->extradata_size = sh_audio->wf->cbSize; |
32105 | 141 memcpy(lavc_context->extradata, sh_audio->wf + 1, |
7940 | 142 lavc_context->extradata_size); |
143 } | |
144 | |
14633 | 145 // for QDM2 |
146 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) | |
147 { | |
148 lavc_context->extradata = av_malloc(sh_audio->codecdata_len); | |
149 lavc_context->extradata_size = sh_audio->codecdata_len; | |
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150 memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, |
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151 lavc_context->extradata_size); |
14633 | 152 } |
153 | |
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154 /* open it */ |
34530 | 155 if (avcodec_open2(lavc_context, lavc_codec, &opts) < 0) { |
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156 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); |
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157 return 0; |
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158 } |
34523 | 159 av_dict_free(&opts); |
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160 mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); |
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161 |
8101 | 162 // printf("\nFOURCC: 0x%X\n",sh_audio->format); |
163 if(sh_audio->format==0x3343414D){ | |
164 // MACE 3:1 | |
165 sh_audio->ds->ss_div = 2*3; // 1 samples/packet | |
35257 | 166 sh_audio->ds->ss_mul = sh_audio->wf ? 2*sh_audio->wf->nChannels : 2; // 1 byte*ch/packet |
8101 | 167 } else |
168 if(sh_audio->format==0x3643414D){ | |
169 // MACE 6:1 | |
170 sh_audio->ds->ss_div = 2*6; // 1 samples/packet | |
35257 | 171 sh_audio->ds->ss_mul = sh_audio->wf ? 2*sh_audio->wf->nChannels : 2; // 1 byte*ch/packet |
8101 | 172 } |
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173 |
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174 // Decode at least 1 byte: (to get header filled) |
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175 do { |
29512 | 176 x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); |
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177 } while (x <= 0 && tries++ < 5); |
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178 if(x>0) sh_audio->a_buffer_len=x; |
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179 |
5480 | 180 sh_audio->i_bps=lavc_context->bit_rate/8; |
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181 if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec) |
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182 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; |
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183 |
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184 switch (lavc_context->sample_fmt) { |
35134 | 185 case AV_SAMPLE_FMT_U8: case AV_SAMPLE_FMT_U8P: |
186 case AV_SAMPLE_FMT_S16: case AV_SAMPLE_FMT_S16P: | |
187 case AV_SAMPLE_FMT_S32: case AV_SAMPLE_FMT_S32P: | |
188 case AV_SAMPLE_FMT_FLT: case AV_SAMPLE_FMT_FLTP: | |
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189 break; |
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190 default: |
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191 return 0; |
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192 } |
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193 setup_format(sh_audio, sh_audio->context); |
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194 return 1; |
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195 } |
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196 |
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197 static void uninit(sh_audio_t *sh) |
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198 { |
7940 | 199 AVCodecContext *lavc_context = sh->context; |
200 | |
201 if (avcodec_close(lavc_context) < 0) | |
5447 | 202 mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec); |
36369 | 203 av_freep(&lavc_context->opaque); |
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204 av_freep(&lavc_context->extradata); |
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205 av_freep(&lavc_context); |
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206 } |
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207 |
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208 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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209 { |
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210 AVCodecContext *lavc_context = sh->context; |
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211 switch(cmd){ |
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212 case ADCTRL_RESYNC_STREAM: |
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213 avcodec_flush_buffers(lavc_context); |
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214 ds_clear_parser(sh->ds); |
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215 return CONTROL_TRUE; |
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216 } |
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217 return CONTROL_UNKNOWN; |
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218 } |
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219 |
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220 static av_always_inline void copy_samples_planar(size_t bps, |
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221 size_t nb_samples, |
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222 size_t nb_channels, |
35134 | 223 unsigned char *dst, |
224 unsigned char **src) | |
225 { | |
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226 size_t s, c, o = 0; |
35134 | 227 |
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228 #if HAVE_NEON |
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229 if (nb_channels == 2 && bps == 4) { |
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230 const unsigned char *src0 = src[0]; |
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231 const unsigned char *src1 = src[1]; |
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232 size_t aligned = nb_samples & ~7; |
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233 const unsigned char *src0_end = src0 + aligned*bps; |
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234 while (src0 < src0_end) { |
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235 __asm__ ( |
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236 "vld1.32 {q0}, [%0]!\n\t" |
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237 "vld1.32 {q1}, [%1]!\n\t" |
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238 "vld1.32 {q2}, [%0]!\n\t" |
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239 "vld1.32 {q3}, [%1]!\n\t" |
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240 "vst2.32 {q0,q1}, [%2]!\n\t" |
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241 "vst2.32 {q2,q3}, [%2]!\n\t" |
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242 : "+&r"(src0), "+&r"(src1), "+&r"(dst) |
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243 :: "q0", "q1", "q2", "q3", "memory"); |
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244 } |
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245 o += aligned*bps; |
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246 nb_samples -= aligned; |
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247 } else if (nb_channels == 2 && bps == 2) { |
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248 const unsigned char *src0 = src[0]; |
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249 const unsigned char *src1 = src[1]; |
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250 size_t aligned = nb_samples & ~15; |
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251 const unsigned char *src0_end = src0 + aligned*bps; |
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252 while (src0 < src0_end) { |
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253 __asm__ ( |
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254 "vld1.16 {q0}, [%0]!\n\t" |
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255 "vld1.16 {q1}, [%1]!\n\t" |
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256 "vld1.16 {q2}, [%0]!\n\t" |
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257 "vld1.16 {q3}, [%1]!\n\t" |
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258 "vst2.16 {q0,q1}, [%2]!\n\t" |
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259 "vst2.16 {q2,q3}, [%2]!\n\t" |
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260 : "+&r"(src0), "+&r"(src1), "+&r"(dst) |
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261 :: "q0", "q1", "q2", "q3", "memory"); |
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262 } |
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263 o += aligned*bps; |
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264 nb_samples -= aligned; |
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265 } |
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266 #endif |
35134 | 267 for (s = 0; s < nb_samples; s++) { |
268 for (c = 0; c < nb_channels; c++) { | |
269 memcpy(dst, src[c] + o, bps); | |
270 dst += bps; | |
271 } | |
272 o += bps; | |
273 } | |
274 } | |
275 | |
276 static int copy_samples(AVCodecContext *avc, AVFrame *frame, | |
277 unsigned char *buf, int max_size) | |
278 { | |
279 int channels = avc->channels; | |
280 int sample_size = av_get_bytes_per_sample(avc->sample_fmt); | |
281 int size = channels * sample_size * frame->nb_samples; | |
282 | |
283 if (size > max_size) { | |
284 av_log(avc, AV_LOG_ERROR, | |
285 "Buffer overflow while decoding a single frame\n"); | |
286 return AVERROR(EINVAL); /* same as avcodec_decode_audio3 */ | |
287 } | |
288 /* TODO reorder channels at the same time */ | |
289 if (av_sample_fmt_is_planar(avc->sample_fmt)) { | |
290 switch (sample_size) { | |
291 case 1: | |
292 copy_samples_planar(1, frame->nb_samples, channels, | |
293 buf, frame->extended_data); | |
294 break; | |
295 case 2: | |
296 copy_samples_planar(2, frame->nb_samples, channels, | |
297 buf, frame->extended_data); | |
298 break; | |
299 case 4: | |
300 copy_samples_planar(4, frame->nb_samples, channels, | |
301 buf, frame->extended_data); | |
35342 | 302 break; |
35134 | 303 default: |
304 copy_samples_planar(sample_size, frame->nb_samples, channels, | |
305 buf, frame->extended_data); | |
306 } | |
307 } else { | |
308 memcpy(buf, frame->data[0], size); | |
309 } | |
310 return size; | |
311 } | |
312 | |
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313 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) |
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314 { |
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315 unsigned char *start=NULL; |
35134 | 316 int y,len=-1, got_frame; |
36771 | 317 AVFrame *frame = av_frame_alloc(); |
35134 | 318 |
319 if (!frame) | |
320 return AVERROR(ENOMEM); | |
321 | |
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322 while(len<minlen){ |
29913 | 323 AVPacket pkt; |
23456 | 324 int len2=maxlen; |
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325 double pts; |
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326 int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); |
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327 if(x<=0) { |
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328 start = NULL; |
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329 x = 0; |
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330 ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0); |
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331 if (x <= 0) |
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332 break; // error |
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333 } else { |
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334 int in_size = x; |
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335 int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0); |
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336 sh_audio->ds->buffer_pos -= in_size - consumed; |
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337 } |
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338 |
29913 | 339 av_init_packet(&pkt); |
340 pkt.data = start; | |
341 pkt.size = x; | |
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342 if (pts != MP_NOPTS_VALUE) { |
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343 sh_audio->pts = pts; |
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344 sh_audio->pts_bytes = 0; |
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345 } |
35134 | 346 y=avcodec_decode_audio4(sh_audio->context, frame, &got_frame, &pkt); |
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flushing stuff after seeking (finally we can view MPEG without thouse blocks after seeking with -vc ffmpeg12)
michael
parents:
9547
diff
changeset
|
347 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout); |
32496
5376d7337fcf
Handle EAGAIN in ad_ffmpeg, this is necessary for LATM to work properly with
reimar
parents:
32390
diff
changeset
|
348 // LATM may need many packets to find mux info |
5376d7337fcf
Handle EAGAIN in ad_ffmpeg, this is necessary for LATM to work properly with
reimar
parents:
32390
diff
changeset
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349 if (y == AVERROR(EAGAIN)) |
5376d7337fcf
Handle EAGAIN in ad_ffmpeg, this is necessary for LATM to work properly with
reimar
parents:
32390
diff
changeset
|
350 continue; |
5340
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
351 if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } |
30272
c6c40936049c
We only need to disable seeking back in ad_ffmpeg when we actually _use_
reimar
parents:
30088
diff
changeset
|
352 if(!sh_audio->parser && y<x) |
30088
4977e04f3a18
Add support for parsing audio streams (though should be easy to extend to video)
reimar
parents:
29928
diff
changeset
|
353 sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) |
35134 | 354 if (!got_frame) |
355 continue; | |
356 len2 = copy_samples(sh_audio->context, frame, buf, maxlen); | |
357 if (len2 < 0) | |
358 return len2; | |
5340
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
359 if(len2>0){ |
25315
dfa8a510c81c
Fix all current known multi-channel wrong order problems by adding
ulion
parents:
24226
diff
changeset
|
360 if (((AVCodecContext *)sh_audio->context)->channels >= 5) { |
33934
14a1ae73b01d
Updated usage of some deprecated functions in FFmpeg audio decoder.
reimar
parents:
33871
diff
changeset
|
361 int samplesize = av_get_bytes_per_sample(((AVCodecContext *) |
14a1ae73b01d
Updated usage of some deprecated functions in FFmpeg audio decoder.
reimar
parents:
33871
diff
changeset
|
362 sh_audio->context)->sample_fmt); |
29779
282ea4fbe87d
ad_ffmpeg: Fix channel layout for ffvorbis and ffaac
tack
parents:
29667
diff
changeset
|
363 reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, |
25315
dfa8a510c81c
Fix all current known multi-channel wrong order problems by adding
ulion
parents:
24226
diff
changeset
|
364 AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, |
dfa8a510c81c
Fix all current known multi-channel wrong order problems by adding
ulion
parents:
24226
diff
changeset
|
365 ((AVCodecContext *)sh_audio->context)->channels, |
29491
99eda963d27a
Fix incorrect channel ordering for lavc audio codecs (specifically ffac3,
tack
parents:
29263
diff
changeset
|
366 len2 / samplesize, samplesize); |
25315
dfa8a510c81c
Fix all current known multi-channel wrong order problems by adding
ulion
parents:
24226
diff
changeset
|
367 } |
5340
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
368 //len=len2;break; |
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
369 if(len<0) len=len2; else len+=len2; |
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
370 buf+=len2; |
23456 | 371 maxlen -= len2; |
18242
caac2ca98168
4 - Implement a better way to calculate current audio pts and use it for
rtognimp
parents:
18150
diff
changeset
|
372 sh_audio->pts_bytes += len2; |
5340
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
373 } |
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
374 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); |
31671
4541f1921482
Add support for parameter changes (e.g. channel count) during playback.
reimar
parents:
31220
diff
changeset
|
375 |
4541f1921482
Add support for parameter changes (e.g. channel count) during playback.
reimar
parents:
31220
diff
changeset
|
376 if (setup_format(sh_audio, sh_audio->context)) |
4541f1921482
Add support for parameter changes (e.g. channel count) during playback.
reimar
parents:
31220
diff
changeset
|
377 break; |
5340
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
378 } |
35134 | 379 |
36771 | 380 av_frame_free(&frame); |
5340
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
381 return len; |
0f12fb7c1c5d
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
arpi
parents:
diff
changeset
|
382 } |