遇到和这个人一毛一样的问题 声音中存在噪声 其他的还好
[Libav-user] aac encoder in real time scenario
Gerard C.L. gerardcl at gmail.comFri Mar 15 09:19:26 CET 2013
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Good moring, I've seen that it's necessary to show the init methods, so here you have: ----------------------------------------8<------------------------------------------- int audio_avcodec_init_encode(struct audio_avcodec_encode_state *aavces, int bit_rate, int sample_rate, int channels){ int enabled=0; avcodec_register_all(); aavces->c= NULL; /* find the encoder */ aavces->codec = avcodec_find_encoder(CODEC_ID_AAC); //AQUÍ STRING *codec, ara AAC default if (!aavces->codec) { fprintf(stderr, "\n[avcodec - audio - encode] Codec not found"); //exit(1); return enabled; }else enabled = 1; aavces->c= avcodec_alloc_context(); /* put sample parameters */ aavces->c->bit_rate = bit_rate;//64000; aavces->c->sample_fmt = AV_SAMPLE_FMT_S16; //aavces->c->channel_layout = AV_CH_LAYOUT_STEREO; aavces->c->sample_rate = sample_rate;//48000; //TODO: get it from dp_map aavces->c->channels = channels;//2; //TODO aavces->c->profile = FF_PROFILE_AAC_MAIN;//FF_PROFILE_AAC_LOW; //aavces->c->time_base = (AVRational){1, sample_rate}; aavces->c->time_base.num = 1; aavces->c->time_base.den = sample_rate; aavces->c->codec_type = AVMEDIA_TYPE_AUDIO; /* open it */ if (avcodec_open(aavces->c, aavces->codec) < 0) { fprintf(stderr, "\n[avcodec - audio - encode] Could not open codec"); //exit(1); return enabled; }else enabled = 1; /* the codec gives us the frame size, in samples */ //aavces->frame_size = aavces->c->frame_size; //aavces->samples = malloc(aavces->frame_size * 2 * aavces->c->channels); aavces->outbuf_size = 1024;//FF_MIN_BUFFER_SIZE * 10; aavces->outbuf = (uint8_t *)av_malloc(aavces->outbuf_size); aavces->fifo_buf = av_fifo_alloc(2*MAX_AUDIO_PACKET_SIZE);//FF_MIN_BUFFER_SIZE); aavces->fifo_outbuf = (uint8_t *)av_malloc(MAX_AUDIO_PACKET_SIZE); if (!(aavces->outbuf == NULL))enabled = 1; printf("\n[avcodec - audio - encode] Enabled!",enabled); return enabled; } ------------------------------->8------------------------------------------------------ Anyone can help me, please? Hope not being a concept problem... Thanks, -------------------- Gerard C.L. -------------------- 2013/3/14 Gerard C.L. <gerardcl at gmail.com> > Hi all, > > I'm developing an AAC encoder in a real time environment. > > The scene is: > - Capture format -> PCM: 48kHz, stereo, 16b/sample. at 25fps -> so, per > frame, 7680Bytes have to be encoded. > > The first problem become when I realised that the encoder works on fixed > chunk sizes (in this case, for the audio configuration, the size is > 4096Bytes per chunk). So, working like a file encoder, I was only encoding > 4096bytes of the 7680 per frame. > The solution was implementing FIFOs, using the av_fifo_.. methods. So now, > I can hear the entire captured sound per frame, but I hear some garbage and > I don't know if it's because of the encoder or how I work with the fifo or > if I have conceptual errors in my mind. To note that I'm playing the sound > after saving it to a file, could it be also the problem? > > I'm copying the piece of code I've implemented right now, I'd love if some > one gets the error... I'm so noob... > > > -----------------------------------8<------------------------------------------------------------------ > int audio_avcodec_encode(struct audio_avcodec_encode_state *aavces, > unsigned char *inbuf, unsigned char *outbuf, int inbufsize) { > AVPacket pkt; > int frameBytes; > int outsize = 0; > int packetSize = 0; > int ret; > int nfifoBytes; > int encBytes = 0; > int sizeTmp = 0; > > frameBytes = aavces->c->frame_size * aavces->c->channels * 2; > av_fifo_realloc2(aavces->fifo_buf,av_fifo_size(aavces->fifo_buf) + > inbufsize); > > // Put the raw audio samples into the FIFO. > ret = av_fifo_generic_write(aavces->fifo_buf, /*(int8_t*)*/inbuf, > inbufsize, NULL ); > > printf("\n[avcodec encode] raw buffer intput size: %d ; fifo size: > %d",inbufsize, ret); > > //encoding each frameByte block > while ((ret = av_fifo_size(aavces->fifo_buf)) >= frameBytes) { > ret = av_fifo_generic_read(aavces->fifo_buf, > aavces->fifo_outbuf,frameBytes, NULL ); > > av_init_packet(&pkt); > > pkt.size = avcodec_encode_audio(aavces->c, > aavces->outbuf,aavces->outbuf_size, (int16_t*) aavces->fifo_outbuf); > > if (pkt.size < 0) { > printf("FFmpeg : ERROR - Can't encode audio frame."); > } > // Rescale from the codec time_base to the AVStream time_base. > if (aavces->c->coded_frame && aavces->c->coded_frame->pts != > (int64_t) (AV_NOPTS_VALUE )) > pkt.pts = > av_rescale_q(aavces->c->coded_frame->pts,aavces->c->time_base, > aavces->c->time_base); > > printf("\nFFmpeg : (%d) Writing audio frame with PTS: > %lld.",aavces->c->frame_number, pkt.pts); > printf("\n[avcodec - audio - encode] Encoder returned %d bytes of > data",pkt.size); > > pkt.data = aavces->outbuf; > pkt.flags |= AV_PKT_FLAG_KEY; > > memcpy(outbuf, pkt.data, pkt.size); > } > > // any bytes left in audio FIFO to encode? > nfifoBytes = av_fifo_size(aavces->fifo_buf); > > printf("\n[avcodec encode] raw buffer intput size: %d", nfifoBytes); > > if (nfifoBytes > 0) { > memset(aavces->fifo_outbuf, 0, frameBytes); > if (aavces->c->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME) { > int nFrameSizeTmp = aavces->c->frame_size; > if (aavces->c->frame_size != 1 && > (aavces->c->codec->capabilities & CODEC_CAP_SMALL_LAST_FRAME)) > aavces->c->frame_size = nfifoBytes / (aavces->c->channels > * 2); > > if (av_fifo_generic_read(aavces->fifo_buf, > aavces->fifo_outbuf,nfifoBytes, NULL ) == 0) { > if (aavces->c->frame_size != 1) > encBytes = avcodec_encode_audio(aavces->c, > aavces->outbuf,aavces->outbuf_size,(int16_t*) aavces->fifo_outbuf); > else > encBytes = avcodec_encode_audio(aavces->c, > aavces->outbuf,nfifoBytes, (int16_t*) aavces->fifo_outbuf); > } > aavces->c->frame_size = nFrameSizeTmp;// restore the native > frame size > } else > printf("\n[audio encoder] codec does not support small > frames"); > } > > // Now flush the encoder. > if (encBytes <= 0){ > encBytes = avcodec_encode_audio(aavces->c, > aavces->outbuf,aavces->outbuf_size, NULL ); > printf("\nFFmpeg : flushing the encoder"); > } > if (encBytes < 0) { > printf("\nFFmpeg : ERROR - Can't encode LAST audio frame."); > } > av_init_packet(&pkt); > > sizeTmp = pkt.size; > > pkt.size = encBytes; > pkt.data = aavces->outbuf; > pkt.flags |= AV_PKT_FLAG_KEY; > > // Rescale from the codec time_base to the AVStream time_base. > if (aavces->c->coded_frame && aavces->c->coded_frame->pts != (int64_t) > (AV_NOPTS_VALUE )) > pkt.pts = > av_rescale_q(aavces->c->coded_frame->pts,aavces->c->time_base, > aavces->c->time_base); > > printf("\nFFmpeg : (%d) Writing audio frame with PTS: > %lld.",aavces->c->frame_number, pkt.pts); > printf("\n[avcodec - audio - encode] Encoder returned %d bytes of > data\n",pkt.size); > > memcpy(outbuf + sizeTmp, pkt.data, pkt.size); > > outsize = sizeTmp + pkt.size; > > return outsize; > } > > -------------------------------------------------->8------------------------------------------------- > > > Then, I'm saving outbuf with outsize per frame encoded. > > Any idea of what I'm doing wrong? > > Thanks in advance! > -------------------- > Gerard C.L. > -------------------- > -------------- next part -------------- An HTML attachment was scrubbed... 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