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Userland+LibAudio: Make audio applications support dynamic sample rate

All audio applications (aplay, Piano, Sound Player) respect the ability
of the system to have theoretically any sample rate. Therefore, they
resample their own audio into the system sample rate.

LibAudio previously had its loaders resample their own audio, even
though they expose their sample rate. This is now changed. The loaders
output audio data in their file's sample rate, which the user has to
query and resample appropriately. Resampling code from Buffer, WavLoader
and FlacLoader is removed.

Note that these applications only check the sample rate at startup,
which is reasonable (the user has to restart applications when changing
the sample rate). Fully dynamic adaptation could both lead to errors and
will require another IPC interface. This seems to be enough for now.
This commit is contained in:
kleines Filmröllchen 2021-08-19 00:13:26 +02:00 committed by Ali Mohammad Pur
parent 9880a5c481
commit d049626f40
12 changed files with 56 additions and 41 deletions

View file

@ -67,19 +67,11 @@ private:
AK::InputMemoryStream* m_memory_stream;
String m_error_string;
// TODO: We should probably move resampling into the audio server.
//
// It would avoid duplicate resampling code and would allow clients
// to be agnostic of the destination audio device's sample rate.
OwnPtr<ResampleHelper<double>> m_resampler;
u32 m_sample_rate { 0 };
u16 m_num_channels { 0 };
PcmSampleFormat m_sample_format;
size_t m_byte_offset_of_data_samples { 0 };
// FIXME: Get this value from the audio server
int m_device_sample_rate { 44100 };
int m_loaded_samples { 0 };
int m_total_samples { 0 };
};