Hi,
I'm by no means an expert, but just a remark - 8kHz is somewhat low
quality, so maybe that's why the audio sounds awful?
Does it sound better when you try resampling it manually via something
like `ffmpeg -i input.wav -ar 8000 output.wav`?
Best,
Paul
Le 26/08/2021 à 20:55, Baumgarten, Julien a écrit :
Hi guys,
I made a previous post in order to get some help in converting +
resampling 16bit PCM (16k HZ) samples to A-law PCM (8k HZ) samples.
I succeeded in converting with another library than ffmpeg but it works.
I am focusing now on the resampling.
I tried the following source code:
int64_t src_ch_layout =AV_CH_LAYOUT_MONO, dst_ch_layout =AV_CH_LAYOUT_MONO; int src_rate =16000,
dst_rate =8000; uint8_t **src_data =NULL, **dst_data =NULL; int src_nb_channels =0, dst_nb_channels
=0; int src_linesize =0, dst_linesize =0; int src_nb_samples =this->_nbSamplesReceived,
dst_nb_samples; enum AVSampleFormat src_sample_fmt =AV_SAMPLE_FMT_U8, dst_sample_fmt
=AV_SAMPLE_FMT_U8; const char *dst_filename ="/tmp/resample.raw"; FILE *dst_file; int
dst_bufsize; const char *fmt; struct SwrContext *swr_ctx; int ret; dst_file = fopen(dst_filename,
"wb"); if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n",
dst_filename); exit(1); }
swr_ctx = swr_alloc(); if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret =AVERROR(ENOMEM); // goto end; }
/* set in options */ av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0); av_opt_set_int(swr_ctx,
"in_sample_rate", src_rate, 0); av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0); /* set out options */
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0); av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0); /* initialize the resampling context */ if ((ret =
swr_init(swr_ctx)) <0) {
fprintf(stderr, "Failed to initialize the resampling
context\n"); // goto end; }
/* Define nb channels */ src_nb_channels =
av_get_channel_layout_nb_channels(src_ch_layout); dst_nb_channels =
av_get_channel_layout_nb_channels(dst_ch_layout); // Define ouput nb samples
dst_nb_samples = av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP); ret =
av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0); if (ret <0) {
fprintf(stderr, "Could not allocate source samples\n"); //
goto end; }
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize,
dst_nb_channels, dst_nb_samples, dst_sample_fmt, 0); if (ret <0) {
fprintf(stderr, "Could not allocate destination samples\n");
// goto end; }
// Fill source samples buffer with A-law samples unsigned int i =0;
std::for_each(this->_test1.begin(), this->_test1.end(), [this, &src_data, &i](const
uint8_t &data) {
src_data[0][i++] = data; }); /* convert to destination format
*/ ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t
**)src_data, src_nb_samples); if (ret <0) {
fprintf(stderr, "Error while converting\n"); // TODO: handle
error }
dst_bufsize = av_samples_get_buffer_size(&dst_linesize,
dst_nb_channels, ret, dst_sample_fmt, 1); if (dst_bufsize <0) {
fprintf(stderr, "Could not get sample buffer size\n"); //
TODO: handle error }
// Write resampled data into file fwrite(dst_data[0], 1, dst_bufsize,
dst_file); if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) <0) {
fprintf(stderr, "Resampling failed.\n"); // TODO: handle
error }
// Close out file fclose(dst_file); // Release memory if (src_data)
av_freep(&src_data[0]); av_freep(&src_data); if (dst_data) av_freep(&dst_data[0]);
av_freep(&dst_data); swr_free(&swr_ctx);
When dst_rate is equal to src_rate, the output is OK without any noise.
However, when dst_rate is lower than src_rate, the audio is awful with
too much noise.
Did I miss something or am I doing something wrong?
Yours sincerely,
Julien BAUMGARTEN
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