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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