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.baumgar...@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.baumgar...@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 <polochonstr...@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.baumgar...@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@ffmpeg.orghttps://ffmpeg.org/mailman/listinfo/libav-user >> >> To unsubscribe, visit link above, or emaillibav-user-requ...@ffmpeg.org with >> subject "unsubscribe". >> >> _______________________________________________ >> Libav-user mailing list >> Libav-user@ffmpeg.org >> https://ffmpeg.org/mailman/listinfo/libav-user >> >> To unsubscribe, visit link above, or email >> libav-user-requ...@ffmpeg.org with subject "unsubscribe". >> > _______________________________________________ > Libav-user mailing list > Libav-user@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/libav-user > > To unsubscribe, visit link above, or email > libav-user-requ...@ffmpeg.org with subject "unsubscribe". >
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