Merge pull request #147 from music-assistant/OzGav-patch-2

Update tech-info.md

Author: OzGav

docs/faq/tech-info.md

@@ -19,13 +19,13 @@ In general, it is not recommended to turn off the volume normalization because t
 
 If audio is only played from one single source (e.g. Deezer) and that audio source already has normalized its audio files, then its safe to disable normalization in MA. If audio is played from different sources or audio is not normalized at the source, it is highly recommended to leave normalization enabled for the best experience.
 
-NOTE that all audio is analyzed at playback time. If no Integrated loudness measurement is available for an audio source, MA will fallback to a dynamic normalizer which is less accurate but will at least prevent a sudden drop or spike in the volume level. See the settings we provide to finetune the behavior in the settings of the Stream Controller, which you can find in the settings menu under the core controller section.
+NOTE that all audio is analyzed at playback time. If no Integrated loudness measurement is available for an audio source, MA will fallback to a dynamic normalizer which is less accurate but will at least prevent a sudden drop or spike in the volume level. See the settings we provide to finetune the behavior in the settings of the Streamserver, which you can find in the settings menu under the core controller section.
 
 **More technical details**
 
-All tracks are processed internally as raw pcm by Music Assistant. So everything that is played will be first decoded to raw pcm of 32 bits floating point in the sample rate of the origin (unless explicit resampling is enabled/needed, such as when flow mode is enabled), and the gain adjustment is done while extracting the raw source media so the pcm chunks passed into the streaming engine have the gain adjustment applied. In this way there should be enough headroom within the (final) 16 or 24 bits bit depth. If a playback target does not support bit depths higher than 16 bits, dithering will be applied to bring the signal down again to 16 bits without qualitt loss.
+All tracks are processed internally as raw pcm by Music Assistant. So everything that is played will be first decoded to raw pcm of 32 bits floating point in the sample rate of the origin (unless explicit resampling is enabled/needed, such as when flow mode is enabled), and the gain adjustment is done while extracting the raw source media so the pcm chunks passed into the streaming engine have the gain adjustment applied. In this way there should be enough headroom within the (final) 16 or 24 bits bit depth. If a playback target does not support bit depths higher than 16 bits, dithering will be applied to bring the signal down again to 16 bits without quality loss.
 
-All further processing in MA is done at PCM raw audio level, such as the optional equalizer (in the future maybe some room correction or other filters) - if "flow mode" is enabled crossfading is also done on the raw pcm chunks but that will resample all audio into one static sample rate and bit depth to create one "flow" of audio to send to the player.
+All further processing in MA is done at PCM raw audio level, such as the DSP Settings - if "flow mode" is enabled crossfading is also done on the raw pcm chunks but that will resample all audio into one static sample rate and bit depth to create one "flow" of audio to send to the player.
 
 The final part in the chain is that MA needs to send the audio to the player. By default we encode the raw PCM into FLAC because it is lossless while still providing a descent amount of compression. For players that can not handle FLAC very well, or simply to save bandwidth, we provide an option (per player) to encode to MP3 instead.