68 lines
2.4 KiB
C++
68 lines
2.4 KiB
C++
#include "audioworker.h"
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AudioWorker::AudioWorker(DecoderCache* decoder_cache, QObject *parent) :
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AudioRenderWorker(decoder_cache, parent)
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{
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}
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void AudioWorker::FrameToValue(StreamPtr stream, FramePtr frame, NodeValueTable *table)
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{
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Q_UNUSED(stream)
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table->Push(NodeParam::kSamples, frame->ToByteArray());
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}
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void AudioWorker::RunNodeAccelerated(const Node *node, const TimeRange &range, const NodeValueDatabase &input_params_in, NodeValueTable *output_params)
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{
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// Check if node processes samples
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if (!node->ProcessesSamplesFrom()) {
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return;
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}
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// Copy database so we can make some temporary modifications to it
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NodeValueDatabase input_params = input_params_in;
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// Try to find the sample buffer in the table
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QVariant samples_var = input_params[node->ProcessesSamplesFrom()].Get(NodeParam::kSamples);
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// If there isn't one, there's nothing to do
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if (samples_var.isNull()) {
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return;
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}
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QByteArray input_buffer = samples_var.toByteArray();
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QByteArray output_buffer(input_buffer.size(), 0);
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int sample_count = input_buffer.size() / audio_params().bytes_per_sample_per_channel();
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// FIXME: Hardcoded float sample format
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for (int i=0;i<sample_count;i++) {
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// Calculate the exact rational time at this sample
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int sample_out_of_channel = i / audio_params().channel_count();
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double sample_to_second = static_cast<double>(sample_out_of_channel) / static_cast<double>(audio_params().sample_rate());
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rational this_sample_time = rational::fromDouble(range.in().toDouble() + sample_to_second);
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// Update all non-sample and non-footage inputs
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foreach (NodeParam* param, node->parameters()) {
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if (param->type() == NodeParam::kInput
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&& param != node->ProcessesSamplesFrom()) {
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NodeInput* input = static_cast<NodeInput*>(param);
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// If the input isn't keyframing, we don't need to update it unless it's connected, in which case it may change
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if (input->IsConnected() || input->is_keyframing()) {
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input_params.Insert(input, ProcessInput(input, TimeRange(this_sample_time, this_sample_time)));
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}
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}
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}
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node->ProcessSamples(&input_params,
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audio_params(),
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reinterpret_cast<const float*>(input_buffer.constData()),
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reinterpret_cast<float*>(output_buffer.data()),
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i);
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}
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output_params->Push(NodeParam::kSamples, output_buffer);
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}
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