我正在尝试将音频从44Khz重新采样到48Khz，并且在重新采样后得到很小的光噪声。好像有人轻轻地在麦克风上打勾。这是双向发生的。从48Khz到44Khz，反之亦然。
我已经读到，这可能会发生，因为swrContext仍然还有一些数据，而且我应该在重新采样下一帧之前刷新上下文。而且，尽管这会有所帮助（噪音很小），但仍然存在。
我尝试改用FFmpeg重采样滤波器，但是输出只是很大的非相干噪声。我很确定libswresample在重采样时不会输出任何噪音，这意味着我只是不知道如何很好地使用它，而我却缺少一些选项。
这是重采样器的代码。
int ResampleFrame(VideoState * videoState, AVFrame *decoded_audio_frame, enum AVSampleFormat out_sample_fmt, uint8_t * out_buf)
{
int in_sample_rate = videoState->audio->ptrAudioCodecCtx_->sample_rate;
int out_sample_rate = SAMPLE_RATE;
// get an instance of the AudioResamplingState struct, create if NULL
AudioResamplingState* arState = getAudioResampling(videoState->audio->ptrAudioCodecCtx_->channel_layout);
if (!arState->swr_ctx)
{
printf("swr_alloc error.\n");
return -1;
}
// get input audio channels
arState->in_channel_layout = (videoState->audio->ptrAudioCodecCtx_->channels ==
av_get_channel_layout_nb_channels(videoState->audio->ptrAudioCodecCtx_->channel_layout)) ?
videoState->audio->ptrAudioCodecCtx_->channel_layout :
av_get_default_channel_layout(videoState->audio->ptrAudioCodecCtx_->channels);
// check input audio channels correctly retrieved
if (arState->in_channel_layout <= 0)
{
printf("in_channel_layout error.\n");
return -1;
}
arState->out_channel_layout = AV_CH_LAYOUT_STEREO;
// retrieve number of audio samples (per channel)
arState->in_nb_samples = decoded_audio_frame->nb_samples;
if (arState->in_nb_samples <= 0)
{
printf("in_nb_samples error.\n");
return -1;
}
// Set SwrContext parameters for resampling
av_opt_set_int(arState->swr_ctx, "in_channel_layout", arState->in_channel_layout, 0);
av_opt_set_int(arState->swr_ctx, "in_sample_rate", in_sample_rate, 0);
av_opt_set_sample_fmt(arState->swr_ctx, "in_sample_fmt", videoState->audio->ptrAudioCodecCtx_->sample_fmt, 0);
// Set SwrContext parameters for resampling
av_opt_set_int(arState->swr_ctx, "out_channel_layout", arState->out_channel_layout, 0);
av_opt_set_int(arState->swr_ctx, "out_sample_rate", out_sample_rate, 0);
av_opt_set_sample_fmt(arState->swr_ctx, "out_sample_fmt", out_sample_fmt, 0);
// initialize SWR context after user parameters have been set
int ret = swr_init(arState->swr_ctx);
if (ret < 0)
{
printf("Failed to initialize the resampling context.\n");
return -1;
}
// retrieve output samples number taking into account the progressive delay
int64_t delay = swr_get_delay(arState->swr_ctx, videoState->audio->ptrAudioCodecCtx_->sample_rate) + arState->in_nb_samples;
arState->out_nb_samples = av_rescale_rnd(delay, out_sample_rate, in_sample_rate, AV_ROUND_UP );
// check output samples number was correctly rescaled
if (arState->out_nb_samples <= 0)
{
printf("av_rescale_rnd error\n");
return -1;
}
// get number of output audio channels
arState->out_nb_channels = av_get_channel_layout_nb_channels(arState->out_channel_layout);
// allocate data pointers array for arState->resampled_data and fill data
// pointers and linesize accordingly
// check memory allocation for the resampled data was successful
ret = av_samples_alloc_array_and_samples(&arState->resampled_data, &arState->out_linesize, arState->out_nb_channels, arState->out_nb_samples, out_sample_fmt, 0);
if (ret < 0)
{
printf("av_samples_alloc_array_and_samples() error: Could not allocate destination samples.\n");
return -1;
}
if (arState->swr_ctx)
{
// do the actual audio data resampling
// check audio conversion was successful
int ret_num_samples = swr_convert(arState->swr_ctx,arState->resampled_data,arState->out_nb_samples,(const uint8_t**)decoded_audio_frame->data, decoded_audio_frame->nb_samples);
//int ret_num_samples = swr_convert_frame(arState->swr_ctx,arState->resampled_data,arState->out_nb_samples,(const uint8_t**)decoded_audio_frame->data, decoded_audio_frame->nb_samples);
if (ret_num_samples < 0)
{
printf("swr_convert_error.\n");
return -1;
}
// get the required buffer size for the given audio parameters
// check audio buffer size
arState->resampled_data_size = av_samples_get_buffer_size(&arState->out_linesize, arState->out_nb_channels,ret_num_samples,out_sample_fmt,1);
if (arState->resampled_data_size < 0)
{
printf("av_samples_get_buffer_size error.\n");
return -1;
}
} else {
printf("swr_ctx null error.\n");
return -1;
}
// copy the resampled data to the output buffer
memcpy(out_buf, arState->resampled_data[0], arState->resampled_data_size);
// flush the swr context
int delayed = swr_convert(arState->swr_ctx,arState->resampled_data,arState->out_nb_samples,NULL,0);
if (arState->resampled_data)
{
av_freep(&arState->resampled_data[0]);
}
av_freep(&arState->resampled_data);
arState->resampled_data = NULL;
int ret_data_size = arState->resampled_data_size;
return ret_data_size;
}
我也尝试使用here所示的过滤器，但我的输出只是噪声。
这是我的过滤器代码
int ResampleFrame(AVFrame *frame, uint8_t *out_buf)
{
/* Push the decoded frame into the filtergraph */
qint32 ret;
ret = av_buffersrc_add_frame_flags(buffersrc_ctx1, frame, AV_BUFFERSRC_FLAG_KEEP_REF);
if (ret < 0)
{
printf("ResampleFrame: Error adding frame to buffer\n");
// Delete input frame and return null
av_frame_unref(frame);
return 0;
}
//printf("resampling\n");
AVFrame *resampled_frame = av_frame_alloc();
/* Pull filtered frames from the filtergraph */
ret = av_buffersink_get_frame(buffersink_ctx1, resampled_frame);
/* Set the timestamp on the resampled frame */
resampled_frame->best_effort_timestamp = resampled_frame->pts;
if (ret < 0)
{
av_frame_unref(frame);
av_frame_unref(resampled_frame);
return 0;
}
int buffer_size = av_samples_get_buffer_size(NULL, 2,resampled_frame->nb_samples,AV_SAMPLE_FMT_S16,1);
memcpy(out_buf,resampled_frame->data,buffer_size);
//av_frame_unref(frame);
av_frame_unref(resampled_frame);
return buffer_size;
}
QString filter_description1 = "aresample=48000,aformat=sample_fmts=s16:channel_layouts=stereo,asetnsamples=n=1024:p=0";
int InitAudioFilter(AVStream *inputStream)
{
char args[512];
int ret;
const AVFilter *buffersrc = avfilter_get_by_name("abuffer");
const AVFilter *buffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
filter_graph = avfilter_graph_alloc();
const enum AVSampleFormat out_sample_fmts[] = {AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE};
const int64_t out_channel_layouts[] = {AV_CH_LAYOUT_STEREO, -1};
const int out_sample_rates[] = {48000, -1};
snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%" PRIx64,
inputStream->codec->time_base.num, inputStream->codec->time_base.den,
inputStream->codec->sample_rate,
av_get_sample_fmt_name(inputStream->codec->sample_fmt),
inputStream->codec->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx1, buffersrc, "in", args, NULL, filter_graph);
if (ret < 0)
{
printf("InitAudioFilter: Unable to create buffersrc\n");
return -1;
}
ret = avfilter_graph_create_filter(&buffersink_ctx1, buffersink, "out", NULL, NULL, filter_graph);
if (ret < 0)
{
printf("InitAudioFilter: Unable to create buffersink\n");
return ret;
}
// set opt SAMPLE FORMATS
ret = av_opt_set_int_list(buffersink_ctx1, "sample_fmts", out_sample_fmts, -1, AV_OPT_SEARCH_CHILDREN);
if (ret < 0)
{
printf("InitAudioFilter: Cannot set output sample format\n");
return ret;
}
// set opt CHANNEL LAYOUTS
ret = av_opt_set_int_list(buffersink_ctx1, "channel_layouts", out_channel_layouts, -1, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
printf("InitAudioFilter: Cannot set output channel layout\n");
return ret;
}
// set opt OUT SAMPLE RATES
ret = av_opt_set_int_list(buffersink_ctx1, "sample_rates", out_sample_rates, -1, AV_OPT_SEARCH_CHILDREN);
if (ret < 0)
{
printf("InitAudioFilter: Cannot set output sample rate\n");
return ret;
}
/* Endpoints for the filter graph. */
outputs -> name = av_strdup("in");
outputs -> filter_ctx = buffersrc_ctx1;
outputs -> pad_idx = 0;
outputs -> next = NULL;
/* Endpoints for the filter graph. */
inputs -> name = av_strdup("out");
inputs -> filter_ctx = buffersink_ctx1;
inputs -> pad_idx = 0;
inputs -> next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_description1.toStdString().c_str(), &inputs, &outputs, NULL)) < 0)
{
printf("InitAudioFilter: Could not add the filter to graph\n");
}
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
{
printf("InitAudioFilter: Could not configure the graph\n");
}
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
AVFilterLink *outlink = buffersink_ctx1->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
QString str = args;
printf("Output: srate:%dHz fmt:%s chlayout: %s\n", (int) outlink->sample_rate,
av_get_sample_fmt_name((AVSampleFormat) outlink->format),
str.toStdString().c_str());
filterGraphInitialized_ = true;
}
由于我对此没有太多的过滤器或音频经验，因此我可能在这里也缺少一些东西。但是不知道是什么。
谢谢