I am so sorry to disturb you. when i use pre-train soundnet to speech emotion recognition, I have some questions. Could you please give me a hand? Thanks
Question 1:
wav, sr = torchaudio.load(path) reads the audio samples, then it is preprocessed by wav.unsqueeze(1).unsqueeze(-1).repeat(1,1,8,1).
what are the requirements for the audio sample rate? Does the sample rate must be 22050? what are the other restrictions?
Question 2:
the last layer is nn.Conv2d(1024, 401, kernel_size=(8, 1), stride=(2, 1)) to extract speech features.
Feature size varies depending upon the length of the audio, what does it depend upon? I want to use the feature for audio classification. How do I get constant dimension feature vector for all of my audio files?
the same as your mentioned, an audio file with 1476864 samples produces feature of dimension [1x1024x46x1] and other files with 2199168 samples produce a feature of dimension [1x1024x68x1]. [1x1024x46x1], 1 represents batch, 1024 channel_out, what is 46 represented? what is the last dimension 1 represented?
Question 3:
How do get constant dimension feature vector for both files? Finally, when I try to classify, What do I need to do with the features of ouput ( 1, 401, feature, 1)so that I can use them in the final classification task? how can the faltten method be better, (batch, channel_out* 1, feature)? average on the channel? or other methods?
PS
I am new to audio and DL, sorry ask basic problem
Thanks
best
I am so sorry to disturb you. when i use pre-train soundnet to speech emotion recognition, I have some questions. Could you please give me a hand? Thanks
Question 1:
wav, sr = torchaudio.load(path) reads the audio samples, then it is preprocessed by wav.unsqueeze(1).unsqueeze(-1).repeat(1,1,8,1).
what are the requirements for the audio sample rate? Does the sample rate must be 22050? what are the other restrictions?
Question 2:
the last layer is nn.Conv2d(1024, 401, kernel_size=(8, 1), stride=(2, 1)) to extract speech features.
Feature size varies depending upon the length of the audio, what does it depend upon? I want to use the feature for audio classification. How do I get constant dimension feature vector for all of my audio files?
the same as your mentioned, an audio file with 1476864 samples produces feature of dimension [1x1024x46x1] and other files with 2199168 samples produce a feature of dimension [1x1024x68x1]. [1x1024x46x1], 1 represents batch, 1024 channel_out, what is 46 represented? what is the last dimension 1 represented?
Question 3:
How do get constant dimension feature vector for both files? Finally, when I try to classify, What do I need to do with the features of ouput ( 1, 401, feature, 1)so that I can use them in the final classification task? how can the faltten method be better, (batch, channel_out* 1, feature)? average on the channel? or other methods?
PS
I am new to audio and DL, sorry ask basic problem
Thanks
best