大多数厂家摄像机输出的音频流格式都是PCM,有一些场合（比如讲音视频流保存成Ts流）需要将PCM格式转成AAC格式。基本的思路是先解码得到音频帧，再将音频帧编码成AAC格式。编码和解码之间需要添加一个filter。filter起到适配的作用。
首先解码：
AVFrame * decode(AVPacket* sample)
{
int gotframe = ;
AVFrame* frame = av_frame_alloc();
AVFrame *filt_frame = nullptr;
auto length = avcodec_decode_audio4(decoderContext, frame, &gotframe, sample);
frame->pts = frame->pkt_pts;
if(length >= && gotframe != )
{
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_PUSH) < ) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
av_frame_free(&frame);
return nullptr;
}
frame->pts = AV_NOPTS_VALUE; /* pull filtered audio from the filtergraph */
filt_frame = av_frame_alloc();
while () {
int ret = av_buffersink_get_frame_flags(buffersink_ctx, filt_frame, AV_BUFFERSINK_FLAG_NO_REQUEST);
if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if(ret < )
{
av_frame_free(&frame);
av_frame_free(&filt_frame);
return nullptr;
} int64_t frame_pts = AV_NOPTS_VALUE;
if (filt_frame->pts != AV_NOPTS_VALUE) {
startTime = (startTime == AV_NOPTS_VALUE) ? : startTime;
AVRational av_time_base_q;
av_time_base_q.num = ;
av_time_base_q.den = AV_TIME_BASE;
filt_frame->pts = frame_pts =
av_rescale_q(filt_frame->pts, buffersink_ctx->inputs[]->time_base, encoderContext->time_base)
- av_rescale_q(startTime, av_time_base_q, encoderContext->time_base);
}
av_frame_free(&frame);
return filt_frame;
}
}
av_frame_free(&filt_frame);
av_frame_free(&frame);
return nullptr;
}
decode 得到AVFrame 也即音频帧，这个frame是不能做为编码的源要经过filter,原因之一是有些摄像机输出的音频包每个packet是320个字节，AAC每个Packet是1024个字节。
初始化Filter：
int initFilters()
{
char args[];
int ret;
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, - };
static const int out_sample_rates[] = {decoderContext->sample_rate , - };
AVRational time_base = input->time_base;
filter_graph = avfilter_graph_alloc(); /* buffer audio source: the decoded frames from the decoder will be inserted here. */ if (!decoderContext->channel_layout)
decoderContext->channel_layout = av_get_default_channel_layout(decoderContext->channels);
sprintf_s(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%I64x",
time_base.num, time_base.den, decoderContext->sample_rate,
av_get_sample_fmt_name(decoderContext->sample_fmt), decoderContext->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < ) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
return ret;
} /* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < ) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
return ret;
} ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -,
AV_OPT_SEARCH_CHILDREN);
if (ret < ) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
return ret;
} ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -,
AV_OPT_SEARCH_CHILDREN);
if (ret < ) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
return ret;
} ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -,
AV_OPT_SEARCH_CHILDREN);
if (ret < ) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
return ret;
} /* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = ;
outputs->next = NULL; inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = ;
inputs->next = NULL; if ((ret = avfilter_graph_parse_ptr(filter_graph, "anull",
&inputs, &outputs, nullptr)) < )
return ret; if ((ret = avfilter_graph_config(filter_graph, NULL)) < )
return ret; av_buffersink_set_frame_size(buffersink_ctx, );
return ;
}
Filter可以简理解为FIFO（当然实际上不是）输入是解码后的AVFrame,输出是编码的源头。AVFrame 经过Filter以后就可以编码了。
shared_ptr<AVPacket> encode(AVFrame * frame)
{
int gotpacket = ;
shared_ptr<AVPacket> packet((AVPacket*)av_malloc(sizeof(AVPacket)), [&](AVPacket *p){av_free_packet(p);av_freep(&p);});
auto pkt = packet.get();
av_init_packet(pkt);
pkt->data = nullptr;
pkt->size = ;
frame->nb_samples = encoderContext->frame_size;
frame->format = encoderContext->sample_fmt;
frame->channel_layout = encoderContext->channel_layout;
int hr = avcodec_encode_audio2(encoderContext.get(), pkt, frame, &gotpacket);
av_frame_free(&frame);
if(gotpacket)
{
if (pkt->pts != AV_NOPTS_VALUE)
pkt->pts = av_rescale_q(pkt->pts, encoderContext->time_base, output->time_base);
if (pkt->dts != AV_NOPTS_VALUE)
pkt->dts = av_rescale_q(pkt->dts, encoderContext->time_base,output->time_base);
if (pkt->duration > )
pkt->duration = int(av_rescale_q(pkt->duration, encoderContext->time_base, output->time_base));
return packet;
}
return nullptr;
}
实际运用中我们用到了智能指针shared_ptr<AVPacket>，也可以不用。但是要注意内存泄露问题。如果程序运行在多核上，建议AVFilterGraph 中thread设置为1.以上代码久经考验。放心使用。如果有什么问题，可以加群 流媒体/Ffmpeg/音视频 127903734进行交流
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