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simplified and optimized samplebuffer

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This commit is contained in:
itsmattkc
2022-05-09 14:55:51 -07:00
parent 58c4600d04
commit bb0fedabba
47 changed files with 214 additions and 252 deletions
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app/render/audioplaybackcache.cpp
+4 -4
View File
@@ -58,7 +58,7 @@ void AudioPlaybackCache::SetParameters(const AudioParams &params)
emit ParametersChanged();
}
void AudioPlaybackCache::WritePCM(const TimeRange &range, const TimeRangeList &valid_ranges, SampleBufferPtr samples)
void AudioPlaybackCache::WritePCM(const TimeRange &range, const TimeRangeList &valid_ranges, const SampleBuffer &samples)
{
// Ensure if we have enough segments to write this data, creating more if not
qint64 length_diff = params_.time_to_bytes_per_channel(range.out()) - playlist_.GetLength();
@@ -72,7 +72,7 @@ void AudioPlaybackCache::WritePCM(const TimeRange &range, const TimeRangeList &v
TimeRangeList ranges_we_validated;
// Calculate buffer size per channel
qint64 buffer_size_per_channel = samples ? samples->sample_count() * params_.bytes_per_sample_per_channel() : 0;
qint64 buffer_size_per_channel = samples.sample_count() * params_.bytes_per_sample_per_channel();
// Write each valid range to the segments
foreach (const TimeRange& r, valid_ranges) {
@@ -114,7 +114,7 @@ void AudioPlaybackCache::WritePCM(const TimeRange &range, const TimeRangeList &v
if (possible_write_length > 0) {
// Assume `samples` is valid if we're here, or else `buffer_size_per_channel` and
// therefore `possible_write_length` will be 0.
seg_file.write(reinterpret_cast<const char*>(samples->data(i)) + src_offset, possible_write_length);
seg_file.write(reinterpret_cast<const char*>(samples.data(i)) + src_offset, possible_write_length);
}
if (possible_write_length < total_write_length) {
@@ -167,7 +167,7 @@ void AudioPlaybackCache::WriteSilence(const TimeRange &range)
{
// WritePCM will automatically fill non-existent bytes with silence, so we just have to send
// it an empty sample buffer
WritePCM(range, {range}, nullptr);
WritePCM(range, {range}, SampleBuffer());
}
void AudioPlaybackCache::ShiftEvent(const rational &from_in_time, const rational &to_in_time)
app/render/audioplaybackcache.h
+1 -1
View File
@@ -66,7 +66,7 @@ public:
void SetParameters(const AudioParams& params);
void WritePCM(const TimeRange &range, const TimeRangeList &valid_ranges, SampleBufferPtr samples);
void WritePCM(const TimeRange &range, const TimeRangeList &valid_ranges, const SampleBuffer &samples);
void WriteWaveform(const TimeRange &range, const TimeRangeList &valid_ranges, const AudioVisualWaveform *waveform);
app/render/job/samplejob.h
+3 -4
View File
@@ -30,7 +30,6 @@ class SampleJob : public AcceleratedJob {
public:
SampleJob()
{
samples_ = nullptr;
}
SampleJob(const NodeValue& value)
@@ -43,18 +42,18 @@ public:
samples_ = row[from].toSamples();
}
SampleBufferPtr samples() const
const SampleBuffer &samples() const
{
return samples_;
}
bool HasSamples() const
{
return samples_ && samples_->is_allocated();
return samples_.is_allocated();
}
private:
SampleBufferPtr samples_;
SampleBuffer samples_;
};
app/render/previewautocacher.cpp
+1 -1
View File
@@ -207,7 +207,7 @@ void PreviewAutoCacher::AudioRendered()
// WritePCM is tolerant to its buffer being null, it will just write silence instead
viewer_node_->audio_playback_cache()->WritePCM(range,
valid_ranges,
watcher->Get().value<SampleBufferPtr>());
watcher->Get().value<SampleBuffer>());
}
viewer_node_->audio_playback_cache()->WriteWaveform(range, valid_ranges, &waveform);
app/render/renderprocessor.cpp
+24 -28
View File
@@ -200,11 +200,11 @@ void RenderProcessor::Run()
ResolveJobs(sample_val, time);
SampleBufferPtr samples = sample_val.toSamples();
if (samples && ticket_->property("enablewaveforms").toBool()) {
SampleBuffer samples = sample_val.toSamples();
if (samples.is_allocated() && ticket_->property("enablewaveforms").toBool()) {
AudioVisualWaveform vis;
vis.set_channel_count(samples->audio_params().channel_count());
vis.OverwriteSamples(samples, samples->audio_params().sample_rate());
vis.set_channel_count(samples.audio_params().channel_count());
vis.OverwriteSamples(samples, samples.audio_params().sample_rate());
ticket_->setProperty("waveform", QVariant::fromValue(vis));
}
@@ -275,9 +275,8 @@ NodeValueTable RenderProcessor::GenerateBlockTable(const Track *track, const Tim
QVector<Block*> active_blocks = track->BlocksAtTimeRange(range);
// All these blocks will need to output to a buffer so we create one here
SampleBufferPtr block_range_buffer = SampleBuffer::CreateAllocated(audio_params,
audio_params.time_to_samples(range.length()));
block_range_buffer->silence();
SampleBuffer block_range_buffer(audio_params, range.length());
block_range_buffer.silence();
NodeValueTable merged_table;
@@ -292,10 +291,10 @@ NodeValueTable RenderProcessor::GenerateBlockTable(const Track *track, const Tim
// Destination buffer
NodeValueTable table = GenerateTable(b, track->GetValueHintForInput(Track::kBlockInput, track->GetArrayIndexFromBlock(b)),Track::TransformRangeForBlock(b, range_for_block));
SampleBufferPtr samples_from_this_block = table.Take(NodeValue::kSamples).toSamples();
SampleBuffer samples_from_this_block = table.Take(NodeValue::kSamples).toSamples();
ClipBlock *clip_cast = dynamic_cast<ClipBlock*>(b);
if (samples_from_this_block) {
if (samples_from_this_block.is_allocated()) {
// If this is a clip, we might have extra speed/reverse information
if (clip_cast) {
double speed_value = clip_cast->speed();
@@ -303,12 +302,12 @@ NodeValueTable RenderProcessor::GenerateBlockTable(const Track *track, const Tim
if (qIsNull(speed_value)) {
// Just silence, don't think there's any other practical application of 0 speed audio
samples_from_this_block->silence();
samples_from_this_block.silence();
} else if (!qFuzzyCompare(speed_value, 1.0)) {
if (clip_cast->maintain_audio_pitch()) {
AudioProcessor processor;
if (processor.Open(samples_from_this_block->audio_params(), samples_from_this_block->audio_params(), speed_value)) {
if (processor.Open(samples_from_this_block.audio_params(), samples_from_this_block.audio_params(), speed_value)) {
AudioProcessor::Buffer out;
// FIXME: This is not the best way to do this, the TempoProcessor works best
@@ -317,7 +316,7 @@ NodeValueTable RenderProcessor::GenerateBlockTable(const Track *track, const Tim
// well on export (assuming audio is all generated at once on export), but
// users may hear clicks and pops in the audio during preview due to this
// approach.
int r = processor.Convert(samples_from_this_block->to_raw_ptrs(), samples_from_this_block->sample_count(), nullptr);
int r = processor.Convert(samples_from_this_block.to_raw_ptrs().data(), samples_from_this_block.sample_count(), nullptr);
if (r < 0) {
qCritical() << "Failed to change tempo of audio:" << r;
@@ -327,16 +326,13 @@ NodeValueTable RenderProcessor::GenerateBlockTable(const Track *track, const Tim
processor.Convert(nullptr, 0, &out);
if (!out.empty()) {
int nb_samples = out.front().size() * samples_from_this_block->audio_params().bytes_per_sample_per_channel();
int nb_samples = out.front().size() * samples_from_this_block.audio_params().bytes_per_sample_per_channel();
if (nb_samples) {
SampleBufferPtr new_samples = SampleBuffer::Create();
new_samples->set_audio_params(samples_from_this_block->audio_params());
new_samples->set_sample_count(nb_samples);
new_samples->allocate();
SampleBuffer new_samples(samples_from_this_block.audio_params(), nb_samples);
for (int i=0; i<out.size(); i++) {
memcpy(new_samples->data(i), out[i].data(), out[i].size());
memcpy(new_samples.data(i), out[i].data(), out[i].size());
}
samples_from_this_block = new_samples;
@@ -346,20 +342,20 @@ NodeValueTable RenderProcessor::GenerateBlockTable(const Track *track, const Tim
}
} else {
// Multiply time
samples_from_this_block->speed(speed_value);
samples_from_this_block.speed(speed_value);
}
}
if (reversed) {
samples_from_this_block->reverse();
samples_from_this_block.reverse();
}
}
int copy_length = qMin(max_dest_sz, samples_from_this_block->sample_count());
int copy_length = qMin(max_dest_sz, samples_from_this_block.sample_count());
// Copy samples into destination buffer
for (int i=0; i<samples_from_this_block->audio_params().channel_count(); i++) {
block_range_buffer->set(i, samples_from_this_block->data(i), destination_offset, copy_length);
for (int i=0; i<samples_from_this_block.audio_params().channel_count(); i++) {
block_range_buffer.set(i, samples_from_this_block.data(i), destination_offset, copy_length);
}
NodeValueTable::Merge({merged_table, table});
@@ -372,7 +368,7 @@ NodeValueTable RenderProcessor::GenerateBlockTable(const Track *track, const Tim
waveform_info.block = clip_cast;
waveform_info.range = range_for_block - b->in();
if (!(waveform_info.silence = !samples_from_this_block.get())) {
if (!(waveform_info.silence = !samples_from_this_block.is_allocated())) {
// Generate a visual waveform from the samples acquired from this block
AudioVisualWaveform visual_waveform;
visual_waveform.set_channel_count(audio_params.channel_count());
@@ -483,7 +479,7 @@ void RenderProcessor::ProcessVideoFootage(TexturePtr destination, const FootageJ
}
}
void RenderProcessor::ProcessAudioFootage(SampleBufferPtr destination, const FootageJob &stream, const TimeRange &input_time)
void RenderProcessor::ProcessAudioFootage(SampleBuffer &destination, const FootageJob &stream, const TimeRange &input_time)
{
DecoderPtr decoder = ResolveDecoderFromInput(stream.decoder(), Decoder::CodecStream(stream.filename(), stream.audio_params().stream_index()));
@@ -526,9 +522,9 @@ void RenderProcessor::ProcessShader(TexturePtr destination, const Node *node, co
render_ctx_->BlitToTexture(shader, job, destination.get());
}
void RenderProcessor::ProcessSamples(SampleBufferPtr destination, const Node *node, const TimeRange &range, const SampleJob &job)
void RenderProcessor::ProcessSamples(SampleBuffer &destination, const Node *node, const TimeRange &range, const SampleJob &job)
{
if (!job.samples() || !job.samples()->is_allocated()) {
if (!job.samples().is_allocated()) {
return;
}
@@ -536,7 +532,7 @@ void RenderProcessor::ProcessSamples(SampleBufferPtr destination, const Node *no
const AudioParams& audio_params = GetCacheAudioParams();
for (int i=0;i<job.samples()->sample_count();i++) {
for (int i=0;i<job.samples().sample_count();i++) {
// Calculate the exact rational time at this sample
double sample_to_second = static_cast<double>(i) / static_cast<double>(audio_params.sample_rate());
app/render/renderprocessor.h
+4 -4
View File
@@ -46,11 +46,11 @@ protected:
virtual void ProcessVideoFootage(TexturePtr destination, const FootageJob &stream, const rational &input_time) override;
virtual void ProcessAudioFootage(SampleBufferPtr destination, const FootageJob &stream, const TimeRange &input_time) override;
virtual void ProcessAudioFootage(SampleBuffer &destination, const FootageJob &stream, const TimeRange &input_time) override;
virtual void ProcessShader(TexturePtr destination, const Node *node, const TimeRange &range, const ShaderJob& job) override;
virtual void ProcessSamples(SampleBufferPtr destination, const Node *node, const TimeRange &range, const SampleJob &job) override;
virtual void ProcessSamples(SampleBuffer &destination, const Node *node, const TimeRange &range, const SampleJob &job) override;
virtual void ProcessColorTransform(TexturePtr destination, const Node *node, const ColorTransformJob& job) override;
@@ -63,9 +63,9 @@ protected:
return render_ctx_->CreateTexture(p);
}
virtual SampleBufferPtr CreateSampleBuffer(const AudioParams &params, int sample_count) override
virtual SampleBuffer CreateSampleBuffer(const AudioParams &params, int sample_count) override
{
return SampleBuffer::CreateAllocated(params, sample_count);
return SampleBuffer(params, sample_count);
}
virtual void ConvertToReferenceSpace(TexturePtr destination, TexturePtr source, const QString &input_cs) override;