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お問い合わせありがとうございます。
ご質問の目的ですが、CSJコーパスで学習された音響/言語モデルが必要という認識で宜しいでしょうか。
nnet3形式のchain modelで宜しければ、学習済みのモデルが下記ページよりダウンロード可能です。
https://www.hark.jp/download/samples/現行の最新版(HARK3.0)のサンプルファイルがCSJで学習されたモデルとなります。
次のファイルをダウンロードして頂けますでしょうか。
HARK_recog_3.0.0_practice2.zip従来のnnet1形式のモデルに比べ認識性能、処理速度なども向上しておりますので、
今後も互換性のためにnnet1形式のモデルを使用して動作させる事は可能ですが
理由が特になければnnet3形式のモデルへ移行される事を推奨しております。モデルの変更にあたり、デコード時に使用する特徴量は
特徴量13次元 + Delta特徴量13次元 + DeltaPower特徴量 の27次元から
特徴量40次元 に変更となっています。ご了承ください。—-
また、将来的にご自身で学習されたモデルへ入れ替えたいという事でしたら、
HARK Forum内の下記スレッドでKaldi公式の学習レシピ(nnet3用)から変更が
必要な点についての回答が行われていますので宜しければご確認ください。
音響モデルの作成についてまず、GMMの学習に13次元のMSLS特徴量、chain学習に40次元のMSLS特徴量が必要となります。
SpeechRecgnitionClientのFEATURESへの入力をSaveHTKFeatureに繋ぎ変える事でHTK形式の特徴量ファイルが出力されます。
デコード時(音声認識セットのサンプル)は40次元の特徴量のみをSpeechRecgnitionClientにて送信していますが、学習の為には別途13次元の特徴量が必要になりますのでご注意ください。
上記のスレッドを参考にHARKで出力したHTK特徴量をKaldiのark形式に変換して頂いた上で、Kaldiへ入力して学習してください。
我々が提供しているモデルではiVectorに100次元を使用しております。
他の設定は HARK_recog_3.0.0_practice2.zip の kaldi_conf ディレクトリをご参照ください。Notes:
Kaldi公式のレシピでPCMデータから直接作成してしまいますと、
MSLS特徴量には対応していませんのでMFCC特徴量が使われます。
MSLSExtractionの代わりにMFCCExtractionというノードを使用する事も可能ですが、
この場合も特徴量の次元数等の設定については学習時と合わせて頂く必要が御座います。以上、宜しくお願い致します。
If the new question is related to this post, please post in the same thread. Conversely, if the new question is not relevant, please post in a new thread with the appropriate subject. It will probably help people with the same problem to find a solution.
I and we (HARK support team) hope this forum will help you.
Best regards,
I’m sorry, I noticed that there was an error in the command I posted before.
I apologize for the confusion caused by the wrong information.Please try the following command:
In this example, plughw:1,0 is a recording device and plughw:0,0 is a playback device.wios -s -y 0 -d plughw:0,0 -z 0 -a plughw:1,0 -c 8 -i input.wav -o output.wavThe meaning of
-y 0is that the playback device is an ALSA device.
The meaning of-d plughw:0,0is that the playback device isplughw:0,0.
The meaning of-z 0is that the recording device is an ALSA device.
The meaning of-a plughw:1,0is that the recording device isplughw:1,0.
-yoption and-zoption are used instead of-xoption.
Because the recording and playback devices are different (there are two), you need-yand-z. If you can record and play back on one device, eg RASP, use only-x.Best regards,
お問い合わせありがとうございます。
> hark-base
> harkmw
> libharkio3
> libhark-netapi
> hark-core
> harktool5現在、公式がサポートしているアーキテクチャはIntel系のx86_64のみ(*1)ですので
上記は全てソースコードからビルドされているという認識で宜しいでしょうか？
*1) FAQ: What are the supported architectures by HARK?ARM系プロセッサでは
CFLAGSやCXXFLAGS等で指定する最適化オプションで
性能が大きく変わる事が御座います。未設定の場合はご検討ください。以降、CPUに合わせたコンパイラオプションが適切に設定されているという前提で
書かせて頂きます。.bashrc等で下記の環境変数を追記してから試して頂けますでしょうか。
.bashrcに書いた場合、端末(bash)を立ち上げなおすと反映されます。
HARK-Designer上から実行する場合は、hark_designerコマンドを起動する
端末を立ち上げなおしてHARK-Designer自体も再度起動してください。export OPENBLAS_NUM_THREADS=1もし、他にもOpenBLASを使用しているアプリケーションがあり
環境変数の設定を全体(全てのbash)に反映したくないというケースでは
HARKを実行する際に、下記のように実行する事でコマンドのみに反映されます。
この方法の場合、HARK-Designer上から実行すると反映されません。
必ずコマンドラインからHARKを実行して頂く必要が御座います。OPENBLAS_NUM_THREADS=1 harkmw ./<your_network_file>.nHARK3.0より、実行コマンドの名称が
harkmwに変更されましたが
harkmwの部分は従来通りbatchflowと書いても実行できます。
互換性の為にエイリアスとして設定されているので機能は同じです。以上、宜しくお願い致します。
First of all, I will talk from factors other than parameters.
Case.1:
If the overall system speed is slowing, the following causes may be considered. If you execute thread processing in a small matrix whose element size is less than a few thousand, calculation units are excessively divided, and the overhead reduces the speed of the OPENBLAS library.Therefore if you are using HARK 3.x, try setting the environment variable
OPENBLAS_NUM_THREADS=1.For example, set the following line in the
.bashrcfile:export OPENBLAS_NUM_THREADS=1In this case, variables will be effective from the newly opened terminal.
Or you can do the following when you run the network file:
OPENBLAS_NUM_THREADS=1 harkmw ./your_networkfile.nIn this case, variables apply only to the command.
Case.2:
If only the processing speed of HARK is decreasing, the following causes may be considered. Please check the following contents.- Make sure that the known sound signal input to the HarkMsgsStreamFromRos node is not intermittent. For example, if a robot performs TTS, the sound signal must be transmitted not only during speech but also during silence periods.
- Make sure that the sampling rate of the microphone array input matches the sampling rate of the sound signal input to the HarkMsgsStreamFromRos node. If it is difficult to match, please adjust the input of HarkMsgsStreamFromRos node to the sampling rate of the microphone array by MultiDownSampler node. Note that if you lower the sampling rate on the microphone array by MultiDownSampler node, you need to recreate the transfer function.
Case.3:
If the parameters of SemiBlindICA node seems to be the cause, please check the following contents.- Please check the background noise level contained in the microphone array input. The SemiBlindICA node’s IS_ZERO parameter is used to determine input levels of INPUT that do not need to be processed, since it is not necessary to perform estimation processing during periods where there is no sound signal input (silence period) of REFERENCE. Note that this parameter should be set with power in the frequency domain.
- Please save the output of AudioStreamFromMic node and HarkMsgsStreamFromRos node with SaveWavePCM. If you compare the sound signals recorded in the two WAV files, you should see something like this:
The waveform of the audio signal obtained at the HarkMsgsStreamFromRos node should be recorded at the AudioStreamFromMic node with a slight delay. The larger the delay amount, the larger the value of the TAP parameter, but if it is excessively large, the amount of calculation increases. However, if it is slow enough to misunderstand it as a hangup, I think that it is not the influence by this parameter.
Best regards,
Thank you for reporting this error.
And, I appreciate your help confirming the operation of the debug version.It has been confirmed that this error is caused by compiler optimization. Currently, we are preparing a package for publishing with a reduced optimization level as a workaround. It will be released in the next few days.
The version with the bug fixed will be “WIOS 3.0.7.1”. Please wait for it to be posted on “News” page.
Best regards,
I prepared three TSP files recorded at different locations with different microphone arrays. The figure below shows an excerpt of one channel, which you open in Audacity to see the spectrogram.
In the two examples below you will see frequency components other than the original TSP signal. This is due to harmonics or reflected waves. It would be desirable to set the largest volume without such extra information.
The following three examples are TSP files that function without any problems in performance. In other words, it is the limit value of the volume that harmonics and reflected waves are barely visible as shown below. Be careful not to make the line due to harmonics or reflected waves more strongly than that. It is most desirable to adjust to the invisible state as in the example at the top.
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お問い合わせありがとうございます。
DisplayLocalizationノード等へ入力されていますので、PyCodeExecutor3ノードの出力はVector<Source>型になっていなければなりません。全てのノードについて入出力の型が掲載されていますので、詳細はHARK-Documentのノードリファレンスをご参照ください。つまり、今回のケースは出力端子の型設定を
self.outputTypes=("prime_source",)
ではなく
self.outputTypes=("vector_source",)
と設定する必要が御座います。それに伴い
self.outputValues["output"] = self.input[0]
と書かれている行についても次のようにする必要が御座います。
self.outputValues["output"] = [self.input[0]]
更に、入力されたVectorサイズが0である(定位無しの)可能性が御座いますので、次のようにしていなければ実行中にクラッシュする可能性が御座います。
self.outputValues["output"] = [] if len(self.input)<1 else [self.input[0]]また、余談ですが
self.outputValues["output"] = self.input self.outputValues["output"] = self.input[0]のように同一出力端子に2度書き込みを行った場合、
後で書き込まれた値のみが使われます。
calculate()メソッドを抜ける時点で設定した型と
最終的に書き込まれている型が不一致している場合には
エラーが発生しますのでご注意ください。以上、よろしくお願い致します。
> Unfortunately, it did not give me any new information on how to overcome my problem with wios (see results of my wios_check in attachments).
@paul: No. I saw your results and understood at least one of the causes.You can use following channels argument with the wios command. -c <channels> (same as --channels <channels>) *) Please select one of ( 2 ) as <channels>.This means that your device only supports 2 channels. If your device supports 1 or 2 channels, you will see the following message:
You can use following channels argument with the wios command. -c <channels> (same as --channels <channels>) *) Please select one of ( 1-2 ) as <channels>.Also, while wios defaults to 16 kHz for the sampling rate, your device does not support 16 kHz. Your device supports only the sampling rates below. In other words, 44.1 kHz, 48 kHz, etc. should be selected.
your playback device:
You can use following sampling rate argument with the wios command. -f <rate> (same as --frequency <rate>) *) Please select one of ( 44100 48000 96000 192000 ) as <rate>.your capture device:
You can use following sampling rate argument with the wios command. -f <rate> (same as --frequency <rate>) *) Please select one of ( 44100 48000 96000 ) as <rate>.——
You will need to do some things:
– The tsp1.wav that you used is 1 Channel (Mono) data, but you need to make it 2 Chennels (Stereo).
e.g.) If you use a tool calledsox, the following will duplicate the channel after rate conversion:
sox tsp1.wav -r 48000 tsp2.wav remix 1 1
Instead, please try using the output tsp2.wav.
– Please set the channel number explicitly as-c 2on your device. It is not-c 1.
– Please set the sampling rate explicitly as (e.g.)-f 48000on your device. It is not-f 16000(This is the wios default if not set).——
The following commands should work on your device.
arecord -d 16 -D plughw:0,0 -f S32_LE -c 2 -r 48000 output.wav
To make it work the same way:
wios -r -x 0 -t 16 -a plughw:0,0 -e 32 -c 2 -f 48000 -o output.wavThe following commands should work on your device.
aplay -D plughw:0,0 tsp.wav
To make it work the same way:
wios -p -x 0 -d plughw:0,0 -i tsp.wav
*) tsp.wav must be 48 kHz, 32-bit (or 16-bit) 2-channel data.The settings for syncing your recording and playback devices in this example are:
wios -s -x 0 -y plughw:0,0 -z plughw:0,0 -e 32 -c 2 -f 48000 -i tsp.wav -o output.wav
wios -s -y 0 -d plughw:0,0 -z 0 -a plughw:0,0 -e 32 -c 2 -f 48000 -i tsp.wav -o output.wavNotes:
If you get an error message about buffers (for example, buffer overruns), you can work around by setting the buffer size larger than the initial value using the-Nand-Zoptions.I hope this answer will help you solve your problem.
Best regards,
@paul: The upload failed due to a problem with the file permission. Thank you for contacting me.
———-
For those who visited later …This post was corrected on 27th June 2019.
Since an error was found in the attached file, uploading is done again.
Please downloadwios-check2.zip.Thank you for your inquiry.
wios uses ALSA directly.
First, make sure you can record and playback using
arecordoraplay. If you set the number of bits or encoding that the hardware does not support, it may not work properly. In other words, be aware of the default settings that is used by wios.
Secondly, wios implements only some of the features of ALSA, so some devices may not be supported. For example, when recording 24-bit PCM with the--encoding 24option,S24_LEis supported butS24_3LEis not.I created a tool that I attached in this post, which can be used do a quick check. But since it has not been tested thoroughly, there may be some problems remaining. But if it works well, the output result may be helpful.
How to compile:
g++ wios-check.cpp -lasound -o wios-checkHow to use:
./wios-check <device> <type>
<type>is one ofplayback,captureorboth.
defaults:<device>ishw:0,0selected, and<type>isbothselected.
e.g.)./wios-check plughw:0,0 playbackFinally, HARKTOOL5 offers a new way to create transfer functions from TSP recordings that have not been synchronized. The information is written below, so I hope you find it useful.
[HARKTOOL5-GUI documentation] => [Transfer Function Estimation Using Complex Regression Model]
Best regards,
By Sylvia, did you mean you are using Linux Mint 18.3 with Ubuntu Xenial as the package base?
Although Linux Mint was derived from Ubuntu, we do not support distribution for such, So if you are using Sylvia via apt repository for the Ubuntu Xenial, I can advice you to please register again the apt repository by changing the part of$(lsb_release -cs)included in the following command toxenial.sudo bash -c 'echo -e "deb http://archive.hark.jp/harkrepos $(lsb_release -cs) non-free\ndeb-src http://archive.hark.jp/harkrepos $(lsb_release -cs) non-free" > /etc/apt/sources.list.d/hark.list'Previously registered URL (domain) of the apt repository will not be able to connect due to our server migration which may have been causing the error.
Installation Instruction for LinuxBest regards,
