[Libav-user] Resampling = noise
Bob Kirnum
bkirnum at gmail.com
Fri Aug 27 16:31:21 EEST 2021
My understanding is that re-sampling is done on linear PCM, not encoded
audio (G.711 PCM). I always encode after re-sampling.
On Fri, Aug 27, 2021 at 6:38 AM Baumgarten, Julien <
julien.baumgarten at viadialog.com> wrote:
> I don't succeed in encode with ffmpeg library.
> So I do the encoding before the resampling.
>
> [image: avatar] [image: viadialog]
> <https://www.viadialog.com/?utm_source=signature&utm_medium=email&utm_campaign=email_signature_logo>
> 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>
>
> 152 Boulevard Pereire, 75017 Paris
> [image: facebook] <https://www.facebook.com/viadialog>
> [image: twitter] <https://twitter.com/viadialog>
> [image: linkedin] <https://www.linkedin.com/company/viatelecom>
>
> <https://www.viadialog.com/?utm_source=signature&utm_medium=email&utm_campaign=email_signature_banner>
> This email message (including its attachments) is confidential and may
> contain privileged information and is intended solely for the use of the
> individual and/or entity to whom it is addressed. If you are not the
> intended recipient of this e-mail you may not share, distribute or copy
> this e-mail (including its attachments), or any part thereof. If this
> e-mail is received in error, please notify the sender immediately by return
> e-mail and make sure that this e-mail (including its attachments), and all
> copies thereof, are immediately deleted from your system. Please further
> note that when you communicate with us via email or visit our website we
> process your personal data. See our privacy policy for more information
> about how we process it: https://www.viadialog.com/mentions-legales
>
>
> Le ven. 27 août 2021 à 12:55, Bob Kirnum <bkirnum at gmail.com> a écrit :
>
>> 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
>>>
>>>
>>> [image: avatar] [image: viadialog]
>>> <https://www.viadialog.com/?utm_source=signature&utm_medium=email&utm_campaign=email_signature_logo>
>>> 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>
>>>
>>> 152 Boulevard Pereire, 75017 Paris
>>> [image: facebook] <https://www.facebook.com/viadialog>
>>> [image: twitter] <https://twitter.com/viadialog>
>>> [image: linkedin] <https://www.linkedin.com/company/viatelecom>
>>>
>>> <https://www.viadialog.com/?utm_source=signature&utm_medium=email&utm_campaign=email_signature_banner>
>>> This email message (including its attachments) is confidential and may
>>> contain privileged information and is intended solely for the use of the
>>> individual and/or entity to whom it is addressed. If you are not the
>>> intended recipient of this e-mail you may not share, distribute or copy
>>> this e-mail (including its attachments), or any part thereof. If this
>>> e-mail is received in error, please notify the sender immediately by return
>>> e-mail and make sure that this e-mail (including its attachments), and all
>>> copies thereof, are immediately deleted from your system. Please further
>>> note that when you communicate with us via email or visit our website we
>>> process your personal data. See our privacy policy for more information
>>> about how we process it: https://www.viadialog.com/mentions-legales
>>>
>>>
>>> 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]
>>>> <https://www.viadialog.com/?utm_source=signature&utm_medium=email&utm_campaign=email_signature_logo>
>>>>
>>>>
>>>> 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>
>>>>
>>>> 152 Boulevard Pereire, 75017 Paris
>>>> [image: facebook] <https://www.facebook.com/viadialog>
>>>> [image: twitter] <https://twitter.com/viadialog>
>>>> [image: linkedin] <https://www.linkedin.com/company/viatelecom>
>>>>
>>>>
>>>> <https://www.viadialog.com/?utm_source=signature&utm_medium=email&utm_campaign=email_signature_banner>
>>>>
>>>> This email message (including its attachments) is confidential and may
>>>> contain privileged information and is intended solely for the use of the
>>>> individual and/or entity to whom it is addressed. If you are not the
>>>> intended recipient of this e-mail you may not share, distribute or copy
>>>> this e-mail (including its attachments), or any part thereof. If this
>>>> e-mail is received in error, please notify the sender immediately by return
>>>> e-mail and make sure that this e-mail (including its attachments), and all
>>>> copies thereof, are immediately deleted from your system. Please further
>>>> note that when you communicate with us via email or visit our website we
>>>> process your personal data. See our privacy policy for more information
>>>> about how we process it: https://www.viadialog.com/mentions-legales
>>>>
>>>> _______________________________________________
>>>> Libav-user mailing listLibav-user at ffmpeg.orghttps://ffmpeg.org/mailman/listinfo/libav-user
>>>>
>>>> To unsubscribe, visit link above, or emaillibav-user-request at ffmpeg.org with subject "unsubscribe".
>>>>
>>>> _______________________________________________
>>>> Libav-user mailing list
>>>> Libav-user at ffmpeg.org
>>>> https://ffmpeg.org/mailman/listinfo/libav-user
>>>>
>>>> To unsubscribe, visit link above, or email
>>>> libav-user-request at ffmpeg.org with subject "unsubscribe".
>>>>
>>> _______________________________________________
>>> Libav-user mailing list
>>> Libav-user at ffmpeg.org
>>> https://ffmpeg.org/mailman/listinfo/libav-user
>>>
>>> To unsubscribe, visit link above, or email
>>> libav-user-request at ffmpeg.org with subject "unsubscribe".
>>>
>> _______________________________________________
>> Libav-user mailing list
>> Libav-user at ffmpeg.org
>> https://ffmpeg.org/mailman/listinfo/libav-user
>>
>> To unsubscribe, visit link above, or email
>> libav-user-request at ffmpeg.org with subject "unsubscribe".
>>
> _______________________________________________
> Libav-user mailing list
> Libav-user at ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/libav-user
>
> To unsubscribe, visit link above, or email
> libav-user-request at ffmpeg.org with subject "unsubscribe".
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://ffmpeg.org/pipermail/libav-user/attachments/20210827/e58f1006/attachment.htm>
More information about the Libav-user
mailing list