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Fixed audio resampling to now use JUCE instead of FFmpeg. Removed lots of debug code.

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This commit is contained in:
Jonathan Thomas
2012-07-30 11:05:36 -05:00
parent 0aac7871a3
commit 80ca5e8315
2 changed files with 15 additions and 92 deletions
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50
src/Frame.cpp
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@@ -255,30 +255,15 @@ float* Frame::GetInterleavedAudioSamples(int new_sample_rate, int* sample_count)
{
float *output = NULL;
AudioSampleBuffer *buffer = audio;
AudioSampleBuffer *resampled_buffer = NULL;
int num_of_channels = audio->getNumChannels();
int num_of_samples = audio->getNumSamples();
// DEBUG
cout << "output size 1: " << (num_of_channels * num_of_samples) << endl;
cout << "buffer channels 1: " << buffer->getNumChannels() << endl;
cout << "buffer samples 1: " << buffer->getNumSamples() << endl;
cout << "--------------" << endl;
// // Loop through samples in each channel (combining them)
// DEBUG CODE
// if (number == 1)
// {
// cout << "BEFORE RESAMPLE" << endl << endl;
// for (int sample = 0; sample < num_of_samples; sample++)
// {
// for (int channel = 0; channel < num_of_channels; channel++)
// {
// // Add sample to output array
// cout << buffer->getSampleData(channel)[sample] << endl;
// }
// }
// }
// for (int s = 0; s < num_of_samples; s++)
// for (int c = 0; c < num_of_channels; c++)
// cout << buffer->getSampleData(c)[s] << endl;
// Resample to new sample rate (if needed)
if (new_sample_rate != sample_rate)
@@ -295,35 +280,26 @@ float* Frame::GetInterleavedAudioSamples(int new_sample_rate, int* sample_count)
double source_ratio = double(sample_rate) / double(new_sample_rate);
double dest_ratio = double(new_sample_rate) / double(sample_rate);
int new_num_of_samples = num_of_samples * dest_ratio;
resample_source.setResamplingRatio(dest_ratio);
resample_source.setResamplingRatio(source_ratio);
// Prepare to resample
resample_source.prepareToPlay(num_of_samples, new_sample_rate);
// Create a buffer for the newly resampled data
AudioSampleBuffer resampled_buffer(num_of_channels, new_num_of_samples);
resampled_buffer.clear();
const AudioSourceChannelInfo resample_callback_buffer = {&resampled_buffer, 0, new_num_of_samples};
resampled_buffer = new AudioSampleBuffer(num_of_channels, new_num_of_samples);
const AudioSourceChannelInfo resample_callback_buffer = {resampled_buffer, 0, new_num_of_samples};
resample_callback_buffer.clearActiveBufferRegion();
// Resample the current frame's audio buffer (info the temp callback buffer)
resample_source.getNextAudioBlock(resample_callback_buffer);
// Update buffer pointer to this newly resampled buffer
buffer = &resampled_buffer;
buffer = resampled_buffer;
// Update num_of_samples
num_of_samples = new_num_of_samples;
}
// DEBUG
cout << "output size 2: " << (num_of_channels * num_of_samples) << endl;
cout << "buffer channels 2: " << buffer->getNumChannels() << endl;
cout << "buffer samples 2: " << buffer->getNumSamples() << endl;
// 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;
@@ -337,16 +313,14 @@ float* Frame::GetInterleavedAudioSamples(int new_sample_rate, int* sample_count)
//cout << position << ", " << channel << ", " << sample << endl;
output[position] = buffer->getSampleData(channel)[sample];
if (number == 1)
{
cout << buffer->getSampleData(channel)[sample] << endl;
}
// increment position
position++;
}
}
// Clean up
delete resampled_buffer;
// Update sample count (since it might have changed due to resampling)
*sample_count = num_of_samples;