ad_ffmpeg: prefer codec to container samplerate for ffaac
[mplayer/kovensky.git] / libmpcodecs / ad_ffmpeg.c
blobc954fb2311f7ccf04e30bb7f2b0f6e81a0f99778
1 /*
2 * This file is part of MPlayer.
4 * MPlayer is free software; you can redistribute it and/or modify
5 * it under the terms of the GNU General Public License as published by
6 * the Free Software Foundation; either version 2 of the License, or
7 * (at your option) any later version.
9 * MPlayer is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
14 * You should have received a copy of the GNU General Public License along
15 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
16 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
19 #include <stdio.h>
20 #include <stdlib.h>
21 #include <unistd.h>
23 #include "config.h"
24 #include "mp_msg.h"
26 #include "ad_internal.h"
27 #include "libaf/reorder_ch.h"
29 #include "mpbswap.h"
31 static const ad_info_t info =
33 "FFmpeg/libavcodec audio decoders",
34 "ffmpeg",
35 "Nick Kurshev",
36 "ffmpeg.sf.net",
40 LIBAD_EXTERN(ffmpeg)
42 #define assert(x)
44 #include "libavcodec/avcodec.h"
46 extern int avcodec_initialized;
48 static int preinit(sh_audio_t *sh)
50 sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
51 return 1;
54 static int init(sh_audio_t *sh_audio)
56 int tries = 0;
57 int x;
58 AVCodecContext *lavc_context;
59 AVCodec *lavc_codec;
61 mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
62 if(!avcodec_initialized){
63 avcodec_init();
64 avcodec_register_all();
65 avcodec_initialized=1;
68 lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
69 if(!lavc_codec){
70 mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
71 return 0;
74 lavc_context = avcodec_alloc_context();
75 sh_audio->context=lavc_context;
77 lavc_context->sample_rate = sh_audio->samplerate;
78 lavc_context->bit_rate = sh_audio->i_bps * 8;
79 if(sh_audio->wf){
80 lavc_context->channels = sh_audio->wf->nChannels;
81 lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
82 lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
83 lavc_context->block_align = sh_audio->wf->nBlockAlign;
84 lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
86 lavc_context->request_channels = audio_output_channels;
87 lavc_context->codec_tag = sh_audio->format; //FOURCC
88 lavc_context->codec_type = CODEC_TYPE_AUDIO;
89 lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
91 /* alloc extra data */
92 if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
93 lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
94 lavc_context->extradata_size = sh_audio->wf->cbSize;
95 memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX),
96 lavc_context->extradata_size);
99 // for QDM2
100 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
102 lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
103 lavc_context->extradata_size = sh_audio->codecdata_len;
104 memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
105 lavc_context->extradata_size);
108 /* open it */
109 if (avcodec_open(lavc_context, lavc_codec) < 0) {
110 mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
111 return 0;
113 mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
115 // printf("\nFOURCC: 0x%X\n",sh_audio->format);
116 if(sh_audio->format==0x3343414D){
117 // MACE 3:1
118 sh_audio->ds->ss_div = 2*3; // 1 samples/packet
119 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
120 } else
121 if(sh_audio->format==0x3643414D){
122 // MACE 6:1
123 sh_audio->ds->ss_div = 2*6; // 1 samples/packet
124 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
127 // Decode at least 1 byte: (to get header filled)
128 do {
129 x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
130 } while (x <= 0 && tries++ < 5);
131 if(x>0) sh_audio->a_buffer_len=x;
133 sh_audio->channels=lavc_context->channels;
134 sh_audio->samplerate=lavc_context->sample_rate;
135 sh_audio->i_bps=lavc_context->bit_rate/8;
136 switch (lavc_context->sample_fmt) {
137 case SAMPLE_FMT_U8: sh_audio->sample_format = AF_FORMAT_U8; break;
138 case SAMPLE_FMT_S16: sh_audio->sample_format = AF_FORMAT_S16_NE; break;
139 case SAMPLE_FMT_S32: sh_audio->sample_format = AF_FORMAT_S32_NE; break;
140 case SAMPLE_FMT_FLT: sh_audio->sample_format = AF_FORMAT_FLOAT_NE; break;
141 default:
142 mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
143 return 0;
145 /* If the audio is AAC the container level data may be unreliable
146 * because of SBR handling problems (possibly half real sample rate at
147 * container level). Default AAC decoding with ad_faad has used codec-level
148 * values for a long time without generating complaints so it should be OK.
150 if (sh_audio->wf && lavc_context->codec_id != CODEC_ID_AAC) {
151 // If the decoder uses the wrong number of channels all is lost anyway.
152 // sh_audio->channels=sh_audio->wf->nChannels;
153 if (sh_audio->wf->nSamplesPerSec)
154 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
155 if (sh_audio->wf->nAvgBytesPerSec)
156 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
158 sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
159 return 1;
162 static void uninit(sh_audio_t *sh)
164 AVCodecContext *lavc_context = sh->context;
166 if (avcodec_close(lavc_context) < 0)
167 mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
168 av_freep(&lavc_context->extradata);
169 av_freep(&lavc_context);
172 static int control(sh_audio_t *sh,int cmd,void* arg, ...)
174 AVCodecContext *lavc_context = sh->context;
175 switch(cmd){
176 case ADCTRL_RESYNC_STREAM:
177 avcodec_flush_buffers(lavc_context);
178 ds_clear_parser(sh->ds);
179 return CONTROL_TRUE;
181 return CONTROL_UNKNOWN;
184 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
186 unsigned char *start=NULL;
187 int y,len=-1;
188 while(len<minlen){
189 AVPacket pkt;
190 int len2=maxlen;
191 double pts;
192 int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
193 if(x<=0) {
194 start = NULL;
195 x = 0;
196 ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
197 if (x <= 0)
198 break; // error
199 } else {
200 int in_size = x;
201 int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
202 sh_audio->ds->buffer_pos -= in_size - consumed;
204 av_init_packet(&pkt);
205 pkt.data = start;
206 pkt.size = x;
207 if (pts != MP_NOPTS_VALUE) {
208 sh_audio->pts = pts;
209 sh_audio->pts_bytes = 0;
211 y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt);
212 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
213 if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
214 if(!sh_audio->parser && y<x)
215 sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
216 if(len2>0){
217 if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
218 int samplesize = av_get_bits_per_sample_format(((AVCodecContext *)
219 sh_audio->context)->sample_fmt) / 8;
220 reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
221 AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
222 ((AVCodecContext *)sh_audio->context)->channels,
223 len2 / samplesize, samplesize);
225 //len=len2;break;
226 if(len<0) len=len2; else len+=len2;
227 buf+=len2;
228 maxlen -= len2;
229 sh_audio->pts_bytes += len2;
231 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
233 return len;