Skip to content

Fix garbage in exported audio caused by resampling #5552

New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

Sign up for GitHub

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

Merged
PhysSong merged 2 commits into from
Jul 10, 2020
Merged

Fix garbage in exported audio caused by resampling #5552

Show file tree
Hide file tree
Changes from all commits
Commits
Show all changes
2 commits
Select commit Hold shift + click to select a range
6fbe32c
Let resample return actual number of generated samples
thmueller64 Jul 2, 2020
14cf2cf
Let resample return the actual number of generated samples
thmueller64 Jul 2, 2020
File filter

Filter by extension

Filter by extension
Conversations
Failed to load comments.
Loading
Jump to
Jump to file
Failed to load files.
Loading
Diff view
Diff view
2 changes: 1 addition & 1 deletion include/AudioDevice.h
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -115,7 +115,7 @@ class AudioDevice
const fpp_t _frames );

// resample given buffer from samplerate _src_sr to samplerate _dst_sr
void resample( const surroundSampleFrame * _src,
fpp_t resample( const surroundSampleFrame * _src,
const fpp_t _frames,
surroundSampleFrame * _dst,
const sample_rate_t _src_sr,
Expand Down
11 changes: 5 additions & 6 deletions src/core/audio/AudioDevice.cpp
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -93,10 +93,8 @@ fpp_t AudioDevice::getNextBuffer( surroundSampleFrame * _ab )
// resample if necessary
if( mixer()->processingSampleRate() != m_sampleRate )
{
resample( b, frames, _ab, mixer()->processingSampleRate(),
m_sampleRate );
frames = frames * m_sampleRate /
mixer()->processingSampleRate();
frames = resample( b, frames, _ab, mixer()->processingSampleRate(),
m_sampleRate );
}
else
{
Expand Down Expand Up @@ -184,15 +182,15 @@ void AudioDevice::renamePort( AudioPort * )



void AudioDevice::resample( const surroundSampleFrame * _src,
fpp_t AudioDevice::resample( const surroundSampleFrame * _src,
const fpp_t _frames,
surroundSampleFrame * _dst,
const sample_rate_t _src_sr,
const sample_rate_t _dst_sr )
{
if( m_srcState == NULL )
{
return;
return _frames;
}
m_srcData.input_frames = _frames;
m_srcData.output_frames = _frames;
Expand All @@ -206,6 +204,7 @@ void AudioDevice::resample( const surroundSampleFrame * _src,
printf( "AudioDevice::resample(): error while resampling: %s\n",
src_strerror( error ) );
}
return static_cast<fpp_t>(m_srcData.output_frames_gen);
}


Expand Down