audio_core\hle\source.cpp: Improve accuracy of SourceStatus (#7432)
This commit is contained in:
parent
7638f87f74
commit
bb003c2bd4
@ -298,9 +298,9 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config,
|
||||
b.buffer_id,
|
||||
state.mono_or_stereo,
|
||||
state.format,
|
||||
true,
|
||||
{}, // 0 in u32_dsp
|
||||
false,
|
||||
true, // from_queue
|
||||
0, // play_position
|
||||
false, // has_played
|
||||
});
|
||||
}
|
||||
LOG_TRACE(Audio_DSP, "enqueuing queued {} addr={:#010x} len={} id={}", i,
|
||||
@ -321,7 +321,11 @@ void Source::ParseConfig(SourceConfiguration::Configuration& config,
|
||||
void Source::GenerateFrame() {
|
||||
current_frame.fill({});
|
||||
|
||||
if (state.current_buffer.empty() && !DequeueBuffer()) {
|
||||
if (state.current_buffer.empty()) {
|
||||
// TODO(SachinV): Should dequeue happen at the end of the frame generation?
|
||||
if (DequeueBuffer()) {
|
||||
return;
|
||||
}
|
||||
state.enabled = false;
|
||||
state.buffer_update = true;
|
||||
state.last_buffer_id = state.current_buffer_id;
|
||||
@ -330,8 +334,6 @@ void Source::GenerateFrame() {
|
||||
}
|
||||
|
||||
std::size_t frame_position = 0;
|
||||
|
||||
state.current_sample_number = state.next_sample_number;
|
||||
while (frame_position < current_frame.size()) {
|
||||
if (state.current_buffer.empty() && !DequeueBuffer()) {
|
||||
break;
|
||||
@ -358,7 +360,7 @@ void Source::GenerateFrame() {
|
||||
}
|
||||
// TODO(jroweboy): Keep track of frame_position independently so that it doesn't lose precision
|
||||
// over time
|
||||
state.next_sample_number += static_cast<u32>(frame_position * state.rate_multiplier);
|
||||
state.current_sample_number += static_cast<u32>(frame_position * state.rate_multiplier);
|
||||
|
||||
state.filters.ProcessFrame(current_frame);
|
||||
}
|
||||
@ -409,7 +411,6 @@ bool Source::DequeueBuffer() {
|
||||
|
||||
// the first playthrough starts at play_position, loops start at the beginning of the buffer
|
||||
state.current_sample_number = (!buf.has_played) ? buf.play_position : 0;
|
||||
state.next_sample_number = state.current_sample_number;
|
||||
state.current_buffer_physical_address = buf.physical_address;
|
||||
state.current_buffer_id = buf.buffer_id;
|
||||
state.last_buffer_id = 0;
|
||||
@ -420,8 +421,17 @@ bool Source::DequeueBuffer() {
|
||||
state.input_queue.push(buf);
|
||||
}
|
||||
|
||||
LOG_TRACE(Audio_DSP, "source_id={} buffer_id={} from_queue={} current_buffer.size()={}",
|
||||
source_id, buf.buffer_id, buf.from_queue, state.current_buffer.size());
|
||||
// Because our interpolation consumes samples instead of using an index,
|
||||
// let's just consume the samples up to the current sample number.
|
||||
state.current_buffer.erase(
|
||||
state.current_buffer.begin(),
|
||||
std::next(state.current_buffer.begin(), state.current_sample_number));
|
||||
|
||||
LOG_TRACE(Audio_DSP,
|
||||
"source_id={} buffer_id={} from_queue={} current_buffer.size()={}, "
|
||||
"buf.has_played={}, buf.play_position={}",
|
||||
source_id, buf.buffer_id, buf.from_queue, state.current_buffer.size(), buf.has_played,
|
||||
buf.play_position);
|
||||
return true;
|
||||
}
|
||||
|
||||
|
@ -87,8 +87,8 @@ private:
|
||||
Format format;
|
||||
|
||||
bool from_queue;
|
||||
u32_dsp play_position; // = 0;
|
||||
bool has_played; // = false;
|
||||
u32 play_position; // = 0;
|
||||
bool has_played; // = false;
|
||||
|
||||
private:
|
||||
template <class Archive>
|
||||
@ -136,7 +136,6 @@ private:
|
||||
// Current buffer
|
||||
|
||||
u32 current_sample_number = 0;
|
||||
u32 next_sample_number = 0;
|
||||
PAddr current_buffer_physical_address = 0;
|
||||
AudioInterp::StereoBuffer16 current_buffer = {};
|
||||
|
||||
@ -171,7 +170,6 @@ private:
|
||||
ar& mono_or_stereo;
|
||||
ar& format;
|
||||
ar& current_sample_number;
|
||||
ar& next_sample_number;
|
||||
ar& current_buffer_physical_address;
|
||||
ar& current_buffer;
|
||||
ar& buffer_update;
|
||||
|
Loading…
Reference in New Issue
Block a user