mirror of
https://github.com/mpv-player/mpv
synced 2024-11-14 22:48:35 +01:00
6e9cbdc104
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@29305 b3059339-0415-0410-9bf9-f77b7e298cf2
209 lines
6.9 KiB
C
209 lines
6.9 KiB
C
#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include "config.h"
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#include "mp_msg.h"
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#include "help_mp.h"
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#include "ad_internal.h"
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#include "libaf/reorder_ch.h"
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#include "mpbswap.h"
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static ad_info_t info =
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{
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"FFmpeg/libavcodec audio decoders",
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"ffmpeg",
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"Nick Kurshev",
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"ffmpeg.sf.net",
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""
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};
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LIBAD_EXTERN(ffmpeg)
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#define assert(x)
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#include "libavcodec/avcodec.h"
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extern int avcodec_initialized;
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static int preinit(sh_audio_t *sh)
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{
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sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
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return 1;
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}
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static int init(sh_audio_t *sh_audio)
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{
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int x;
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AVCodecContext *lavc_context;
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AVCodec *lavc_codec;
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mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
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if(!avcodec_initialized){
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avcodec_init();
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avcodec_register_all();
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avcodec_initialized=1;
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}
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lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
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if(!lavc_codec){
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll);
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return 0;
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}
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lavc_context = avcodec_alloc_context();
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sh_audio->context=lavc_context;
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lavc_context->sample_rate = sh_audio->samplerate;
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lavc_context->bit_rate = sh_audio->i_bps * 8;
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if(sh_audio->wf){
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lavc_context->channels = sh_audio->wf->nChannels;
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lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
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lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
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lavc_context->block_align = sh_audio->wf->nBlockAlign;
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lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
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}
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lavc_context->request_channels = audio_output_channels;
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lavc_context->codec_tag = sh_audio->format; //FOURCC
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lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
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/* alloc extra data */
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if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
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lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
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lavc_context->extradata_size = sh_audio->wf->cbSize;
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memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX),
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lavc_context->extradata_size);
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}
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// for QDM2
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if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
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{
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lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
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lavc_context->extradata_size = sh_audio->codecdata_len;
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memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
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lavc_context->extradata_size);
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}
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/* open it */
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if (avcodec_open(lavc_context, lavc_codec) < 0) {
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mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
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return 0;
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}
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mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n");
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// printf("\nFOURCC: 0x%X\n",sh_audio->format);
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if(sh_audio->format==0x3343414D){
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// MACE 3:1
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sh_audio->ds->ss_div = 2*3; // 1 samples/packet
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sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
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} else
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if(sh_audio->format==0x3643414D){
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// MACE 6:1
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sh_audio->ds->ss_div = 2*6; // 1 samples/packet
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sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
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}
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// Decode at least 1 byte: (to get header filled)
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x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
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if(x>0) sh_audio->a_buffer_len=x;
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sh_audio->channels=lavc_context->channels;
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sh_audio->samplerate=lavc_context->sample_rate;
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sh_audio->i_bps=lavc_context->bit_rate/8;
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switch (lavc_context->sample_fmt) {
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case SAMPLE_FMT_U8: sh_audio->sample_format = AF_FORMAT_U8; break;
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case SAMPLE_FMT_S16: sh_audio->sample_format = AF_FORMAT_S16_NE; break;
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case SAMPLE_FMT_S32: sh_audio->sample_format = AF_FORMAT_S32_NE; break;
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case SAMPLE_FMT_FLT: sh_audio->sample_format = AF_FORMAT_FLOAT_NE; break;
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default:
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mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
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return 0;
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}
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if(sh_audio->wf){
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// If the decoder uses the wrong number of channels all is lost anyway.
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// sh_audio->channels=sh_audio->wf->nChannels;
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if (sh_audio->wf->nSamplesPerSec)
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sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
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if (sh_audio->wf->nAvgBytesPerSec)
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sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
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}
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sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
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return 1;
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}
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static void uninit(sh_audio_t *sh)
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{
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AVCodecContext *lavc_context = sh->context;
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if (avcodec_close(lavc_context) < 0)
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mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec);
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av_freep(&lavc_context->extradata);
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av_freep(&lavc_context);
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}
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static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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{
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AVCodecContext *lavc_context = sh->context;
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switch(cmd){
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case ADCTRL_RESYNC_STREAM:
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avcodec_flush_buffers(lavc_context);
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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{
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unsigned char *start=NULL;
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int y,len=-1;
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while(len<minlen){
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int len2=maxlen;
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double pts;
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int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
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if(x<=0) break; // error
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if (pts != MP_NOPTS_VALUE) {
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sh_audio->pts = pts;
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sh_audio->pts_bytes = 0;
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}
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y=avcodec_decode_audio2(sh_audio->context,(int16_t*)buf,&len2,start,x);
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//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
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if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
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if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
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if(len2>0){
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if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
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int src_ch_layout = AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT;
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const char *codec=((AVCodecContext*)sh_audio->context)->codec->name;
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if (!strcasecmp(codec, "ac3")
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|| !strcasecmp(codec, "eac3"))
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src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_AC3_DEFAULT;
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else if (!strcasecmp(codec, "dca"))
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src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_DCA_DEFAULT;
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else if (!strcasecmp(codec, "libfaad")
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|| !strcasecmp(codec, "mpeg4aac"))
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src_ch_layout = AF_CHANNEL_LAYOUT_AAC_DEFAULT;
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else if (!strcasecmp(codec, "liba52"))
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src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_LIBA52_DEFAULT;
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else if (!strcasecmp(codec, "vorbis"))
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src_ch_layout = AF_CHANNEL_LAYOUT_VORBIS_DEFAULT;
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else if (!strcasecmp(codec, "flac"))
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src_ch_layout = AF_CHANNEL_LAYOUT_FLAC_DEFAULT;
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else
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src_ch_layout = AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT;
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reorder_channel_nch(buf, src_ch_layout,
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AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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((AVCodecContext *)sh_audio->context)->channels,
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len2 / 2, 2);
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}
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//len=len2;break;
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if(len<0) len=len2; else len+=len2;
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buf+=len2;
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maxlen -= len2;
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sh_audio->pts_bytes += len2;
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}
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mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
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}
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return len;
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}
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