[Libav-user] Resampling = noise
Bob Kirnum
bkirnum at gmail.com
Fri Aug 27 11:54:58 EEST 2021
In addition to re-sampling from 16 kHz to 8 kHz, are you then encoding the
resulting 8 kHz linear PCM (16 bit?) to G.711 (Alaw?)?
On Fri, Aug 27, 2021 at 2:08 AM Baumgarten, Julien <
julien.baumgarten at viadialog.com> wrote:
> Hi Polochon,
>
> Thx for your answer. I know I'll lose on audio quality by resampling 16kHZ
> to 8kHZ but I need to play the audio on VOIP calls which requires G711
> a-law 8k HZ samples :(
> If I work with your command line, the sound is faaaaaaaaaaar much better.
> No noise at all
>
>
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> Le ven. 27 août 2021 à 00:00, Polochon Street <polochonstreet at gmx.fr> a
> écrit :
>
>> 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
>>
>>
>> [image: avatar] [image: viadialog]
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>>
>>
>> Julien BAUMGARTEN
>>
>> Chef de Projet Développement
>>
>> 01 77 45 30 94
>> <0177453094>
>>
>> julien.baumgarten at viadialog.com
>>
>> www.viadialog.com
>>
>> <https://www.viadialog.com/?utm_source=signature&utm_medium=email&utm_campaign=email_signature_link>
>>
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>>
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