💡 [Audio] Rename TODO(Audio-Philippe) as TODO(Audio)

src/InputStream.cpp

@@ -166,7 +166,7 @@ void InputStream::set_device(RtAudio::DeviceInfo const& info)
     params.deviceId  = info.ID;
     params.nChannels = 1;
     unsigned int nb_frames{512};                         // 512 is a decent value that seems to work well.
-    auto const   sample_rate = info.preferredSampleRate; // TODO(Audio-Philippe) Should we use preferredSampleRate or currentSampleRate?
+    auto const   sample_rate = info.preferredSampleRate; // TODO(Audio) Should we use preferredSampleRate or currentSampleRate?
     _backend.openStream(nullptr, &params, RTAUDIO_FLOAT32, sample_rate, &nb_frames, &audio_input_callback, this);
     _backend.startStream();
 

src/Player.cpp

@@ -86,7 +86,7 @@ void Player::recreate_stream_adapted_to_current_audio_data()
         &_parameters,
         nullptr, // No input stream needed
         RTAUDIO_FLOAT32,
-        _data.sample_rate, // TODO(Audio-Philippe) Resample the audio data to make it match the preferredSampleRate of the device. The current strategy works, unless the device does not support the sample_rate used by our audio data, in which case the audio will be played too slow or too fast.
+        _data.sample_rate, // TODO(Audio) Resample the audio data to make it match the preferredSampleRate of the device. The current strategy works, unless the device does not support the sample_rate used by our audio data, in which case the audio will be played too slow or too fast.
         &nb_frames_per_callback,
         &audio_callback,
         this

src/compute_volume.cpp

@@ -7,7 +7,7 @@ auto compute_volume(std::span<float const> data) -> float
 {
     if (data.empty())
         return 0.f;
-    // TODO(Audio-Philippe) Implement a much smarter loudness computation like https://github.com/klangfreund/LUFSMeter/tree/master
+    // TODO(Audio) Implement a much smarter loudness computation like https://github.com/klangfreund/LUFSMeter/tree/master
     float sum_of_squares{0.f};
     for (float const sample : data)
         sum_of_squares += sample * sample;

src/fourier_transform.cpp

@@ -44,7 +44,7 @@ auto fourier_transform(size_t samples_count, ForEachSample const& for_each_sampl
     auto        fft_output = dj::fft1d(fft_input, dj::fft_dir::DIR_FWD);
     float const delta_between_frequencies{audio_data_sample_rate / static_cast<float>(fft_output.size())}; // The values in the `fft_output` correspond to frequencies between 0 and sample_rate, evenly spaced.
 
-    // TODO(Audio-Philippe) Instead of computing the fft on a signal with many samples, and then resizing it to fit the requested `max_output_frequency_in_hz`, we could reduce it's sample rate before computing the fft, to minimize the number of frequencies that are computed for nothing (since they will be discarded afterwards anyways).
+    // TODO(Audio) Instead of computing the fft on a signal with many samples, and then resizing it to fit the requested `max_output_frequency_in_hz`, we could reduce it's sample rate before computing the fft, to minimize the number of frequencies that are computed for nothing (since they will be discarded afterwards anyways).
 
     // Ignore the redondant information and the frequencies that are above max_frequency_in_hz.
     {