Mercurial > mplayer.hg
view libmpcodecs/ad_ffmpeg.c @ 29881:5797dc687a8e
Add preliminary support for streaming via FFmpeg's URProtocol functions.
Basic playback tested for file and http protocols.
author | reimar |
---|---|
date | Tue, 17 Nov 2009 16:09:17 +0000 |
parents | 282ea4fbe87d |
children | 49f3853e9105 |
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#include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #include "mp_msg.h" #include "help_mp.h" #include "ad_internal.h" #include "libaf/reorder_ch.h" #include "mpbswap.h" static ad_info_t info = { "FFmpeg/libavcodec audio decoders", "ffmpeg", "Nick Kurshev", "ffmpeg.sf.net", "" }; LIBAD_EXTERN(ffmpeg) #define assert(x) #include "libavcodec/avcodec.h" extern int avcodec_initialized; static int preinit(sh_audio_t *sh) { sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE; return 1; } static int init(sh_audio_t *sh_audio) { int tries = 0; int x; AVCodecContext *lavc_context; AVCodec *lavc_codec; mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); if(!avcodec_initialized){ avcodec_init(); avcodec_register_all(); avcodec_initialized=1; } lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll); if(!lavc_codec){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); return 0; } lavc_context = avcodec_alloc_context(); sh_audio->context=lavc_context; lavc_context->sample_rate = sh_audio->samplerate; lavc_context->bit_rate = sh_audio->i_bps * 8; if(sh_audio->wf){ lavc_context->channels = sh_audio->wf->nChannels; lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; lavc_context->block_align = sh_audio->wf->nBlockAlign; lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample; } lavc_context->request_channels = audio_output_channels; lavc_context->codec_tag = sh_audio->format; //FOURCC lavc_context->codec_type = CODEC_TYPE_AUDIO; lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi /* alloc extra data */ if (sh_audio->wf && sh_audio->wf->cbSize > 0) { lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); lavc_context->extradata_size = sh_audio->wf->cbSize; memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX), lavc_context->extradata_size); } // for QDM2 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) { lavc_context->extradata = av_malloc(sh_audio->codecdata_len); lavc_context->extradata_size = sh_audio->codecdata_len; memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, lavc_context->extradata_size); } /* open it */ if (avcodec_open(lavc_context, lavc_codec) < 0) { mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); return 0; } mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n"); // printf("\nFOURCC: 0x%X\n",sh_audio->format); if(sh_audio->format==0x3343414D){ // MACE 3:1 sh_audio->ds->ss_div = 2*3; // 1 samples/packet sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet } else if(sh_audio->format==0x3643414D){ // MACE 6:1 sh_audio->ds->ss_div = 2*6; // 1 samples/packet sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet } // Decode at least 1 byte: (to get header filled) do { x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); } while (x <= 0 && tries++ < 5); if(x>0) sh_audio->a_buffer_len=x; sh_audio->channels=lavc_context->channels; sh_audio->samplerate=lavc_context->sample_rate; sh_audio->i_bps=lavc_context->bit_rate/8; switch (lavc_context->sample_fmt) { case SAMPLE_FMT_U8: sh_audio->sample_format = AF_FORMAT_U8; break; case SAMPLE_FMT_S16: sh_audio->sample_format = AF_FORMAT_S16_NE; break; case SAMPLE_FMT_S32: sh_audio->sample_format = AF_FORMAT_S32_NE; break; case SAMPLE_FMT_FLT: sh_audio->sample_format = AF_FORMAT_FLOAT_NE; break; default: mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n"); return 0; } if(sh_audio->wf){ // If the decoder uses the wrong number of channels all is lost anyway. // sh_audio->channels=sh_audio->wf->nChannels; if (sh_audio->wf->nSamplesPerSec) sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; if (sh_audio->wf->nAvgBytesPerSec) sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; } sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8; return 1; } static void uninit(sh_audio_t *sh) { AVCodecContext *lavc_context = sh->context; if (avcodec_close(lavc_context) < 0) mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec); av_freep(&lavc_context->extradata); av_freep(&lavc_context); } static int control(sh_audio_t *sh,int cmd,void* arg, ...) { AVCodecContext *lavc_context = sh->context; switch(cmd){ case ADCTRL_RESYNC_STREAM: avcodec_flush_buffers(lavc_context); return CONTROL_TRUE; } return CONTROL_UNKNOWN; } static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) { unsigned char *start=NULL; int y,len=-1; while(len<minlen){ int len2=maxlen; double pts; int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); if(x<=0) break; // error if (pts != MP_NOPTS_VALUE) { sh_audio->pts = pts; sh_audio->pts_bytes = 0; } y=avcodec_decode_audio2(sh_audio->context,(int16_t*)buf,&len2,start,x); //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout); if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) if(len2>0){ if (((AVCodecContext *)sh_audio->context)->channels >= 5) { int samplesize = av_get_bits_per_sample_format(((AVCodecContext *) sh_audio->context)->sample_fmt) / 8; const char *codec=((AVCodecContext*)sh_audio->context)->codec->name; reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, ((AVCodecContext *)sh_audio->context)->channels, len2 / samplesize, samplesize); } //len=len2;break; if(len<0) len=len2; else len+=len2; buf+=len2; maxlen -= len2; sh_audio->pts_bytes += len2; } mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); } return len; }