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Many improvements for smooth audio playback when 'time' keyframes are used for time remapping.

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- Added new Frame::ReverseAudio() helper method - for time keyframe support playing backwards audio
- Fixed math rounding error on time keyframe Resampler, to use the rounded source samples value
- Removed some unused args and Frame audio methods
- Reset FrameMapper resample context when non-adjacent frames are requested
- Correctly reverse Frame audio if a time keyframe is present, and reversing audio
- Rewrite Keyframe::IsIncreasing() function, to give accurate direction of time keyframes at any index (special logic to support first and last indexes)
- Fixed Keyframe unit tests - since logic has changed a bit
- Improved time mapping unit tests
- Replace many c-style casts with static_cast
- Added new sine.wav test file
This commit is contained in:
Jonathan Thomas
2023-03-10 01:15:14 -06:00
parent d104664da3
commit 09ce05bbf2
15 changed files with 144 additions and 181 deletions
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103
src/Frame.cpp
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@@ -311,92 +311,25 @@ float Frame::GetAudioSample(int channel, int sample, int magnitude_range)
}
// Get an array of sample data (and optional reverse the sample values)
float* Frame::GetAudioSamples(int channel, bool reverse) {
if (reverse) {
// Copy audio buffer, and reverse channel
juce::AudioBuffer<float> *buffer(audio.get());
buffer->reverse(channel, 0, buffer->getNumSamples());
float* Frame::GetAudioSamples(int channel) {
// return JUCE audio data for this channel
return buffer->getWritePointer(channel);
} else {
// return JUCE audio data for this channel
return audio->getWritePointer(channel);
}
}
// Get a planar array of sample data, using any sample rate
float* Frame::GetPlanarAudioSamples(int new_sample_rate, AudioResampler* resampler, int* sample_count)
{
float *output = NULL;
// Copy audio data
juce::AudioBuffer<float> *buffer(audio.get());
int num_of_channels = audio->getNumChannels();
int num_of_samples = GetAudioSamplesCount();
// Resample to new sample rate (if needed)
if (new_sample_rate != sample_rate)
{
// YES, RESAMPLE AUDIO
resampler->SetBuffer(audio.get(), sample_rate, new_sample_rate);
// Resample data, and return new buffer pointer
buffer = resampler->GetResampledBuffer();
// Update num_of_samples
num_of_samples = buffer->getNumSamples();
}
// INTERLEAVE all samples together (channel 1 + channel 2 + channel 1 + channel 2, etc...)
output = new float[num_of_channels * num_of_samples];
int position = 0;
// Loop through samples in each channel (combining them)
for (int channel = 0; channel < num_of_channels; channel++)
{
for (int sample = 0; sample < num_of_samples; sample++)
{
// Add sample to output array
output[position] = buffer->getReadPointer(channel)[sample];
// increment position
position++;
}
}
// Update sample count (since it might have changed due to resampling)
*sample_count = num_of_samples;
// return combined array
return output;
// return JUCE audio data for this channel
return buffer->getWritePointer(channel);
}
// Get an array of sample data (all channels interleaved together), using any sample rate
float* Frame::GetInterleavedAudioSamples(int new_sample_rate, AudioResampler* resampler, int* sample_count, bool reverse)
float* Frame::GetInterleavedAudioSamples(int* sample_count)
{
float *output = NULL;
// Copy audio data
juce::AudioBuffer<float> *buffer(audio.get());
float *output = NULL;
int num_of_channels = audio->getNumChannels();
int num_of_samples = GetAudioSamplesCount();
if (reverse) {
// Reverse audio samples (if needed)
buffer->reverse(0, buffer->getNumSamples());
}
// Resample to new sample rate (if needed)
if (new_sample_rate != sample_rate && resampler)
{
// YES, RESAMPLE AUDIO
resampler->SetBuffer(audio.get(), sample_rate, new_sample_rate);
// Resample data, and return new buffer pointer
buffer = resampler->GetResampledBuffer();
// Update num_of_samples
num_of_samples = buffer->getNumSamples();
}
// INTERLEAVE all samples together (channel 1 + channel 2 + channel 1 + channel 2, etc...)
output = new float[num_of_channels * num_of_samples];
int position = 0;
@@ -700,8 +633,8 @@ void Frame::Thumbnail(std::string path, int new_width, int new_height, std::stri
mask->invertPixels();
// Get pixels
unsigned char *pixels = (unsigned char *) thumbnail->bits();
const unsigned char *mask_pixels = (const unsigned char *) mask->constBits();
unsigned char *pixels = static_cast<unsigned char *>(thumbnail->bits());
const unsigned char *mask_pixels = static_cast<const unsigned char *>(mask->constBits());
// Convert the mask image to grayscale
// Loop through pixels
@@ -865,6 +798,16 @@ void Frame::ResizeAudio(int channels, int length, int rate, ChannelLayout layout
max_audio_sample = length;
}
// Reverse the audio buffer of this frame (will only reverse a single time, regardless of how many times
// you invoke this method)
void Frame::ReverseAudio() {
if (audio && !audio_reversed) {
// Reverse audio buffer
audio->reverse(0, audio->getNumSamples());
audio_reversed = true;
}
}
// Add audio samples to a specific channel
void Frame::AddAudio(bool replaceSamples, int destChannel, int destStartSample, const float* source, int numSamples, float gainToApplyToSource = 1.0f) {
const std::lock_guard<std::recursive_mutex> lock(addingAudioMutex);
@@ -891,6 +834,9 @@ void Frame::AddAudio(bool replaceSamples, int destChannel, int destStartSample,
// Calculate max audio sample added
if (new_length > max_audio_sample)
max_audio_sample = new_length;
// Reset audio reverse flag
audio_reversed = false;
}
// Apply gain ramp (i.e. fading volume)
@@ -1045,4 +991,7 @@ void Frame::AddAudioSilence(int numSamples)
// Calculate max audio sample added
max_audio_sample = numSamples;
// Reset audio reverse flag
audio_reversed = false;
}