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-rw-r--r--servers/audio/audio_rb_resampler.cpp32
-rw-r--r--servers/audio/audio_stream.cpp16
-rw-r--r--servers/audio/audio_stream.h2
-rw-r--r--servers/audio/effects/audio_effect_filter.h13
-rw-r--r--servers/audio/effects/reverb.cpp24
5 files changed, 41 insertions, 46 deletions
diff --git a/servers/audio/audio_rb_resampler.cpp b/servers/audio/audio_rb_resampler.cpp
index 84a87de2e2..d9b3579812 100644
--- a/servers/audio/audio_rb_resampler.cpp
+++ b/servers/audio/audio_rb_resampler.cpp
@@ -79,53 +79,27 @@ uint32_t AudioRBResampler::_resample(AudioFrame *p_dest, int p_todo, int32_t p_i
p_dest[i] = AudioFrame(v0, v1);
}
- // For now, channels higher than stereo are almost ignored
+ // This will probably never be used, but added anyway
if (C == 4) {
- // FIXME: v2 and v3 are not being used (thus were commented out to prevent
- // compilation warnings, but they should likely be uncommented *and* used).
- // See also C == 6 with similar issues.
float v0 = rb[(pos << 2) + 0];
float v1 = rb[(pos << 2) + 1];
- /*
- float v2 = rb[(pos << 2) + 2];
- float v3 = rb[(pos << 2) + 3];
- */
float v0n = rb[(pos_next << 2) + 0];
float v1n = rb[(pos_next << 2) + 1];
- /*
- float v2n = rb[(pos_next << 2) + 2];
- float v3n = rb[(pos_next << 2) + 3];
- */
-
v0 += (v0n - v0) * frac;
v1 += (v1n - v1) * frac;
- /*
- v2 += (v2n - v2) * frac;
- v3 += (v3n - v3) * frac;
- */
p_dest[i] = AudioFrame(v0, v1);
}
if (C == 6) {
- // FIXME: Lot of unused assignments here, but it seems like intermediate calculations
- // should be done as for C == 2 (C == 4 also has some unused assignments).
float v0 = rb[(pos * 6) + 0];
float v1 = rb[(pos * 6) + 1];
- /*
- float v2 = rb[(pos * 6) + 2];
- float v3 = rb[(pos * 6) + 3];
- float v4 = rb[(pos * 6) + 4];
- float v5 = rb[(pos * 6) + 5];
float v0n = rb[(pos_next * 6) + 0];
float v1n = rb[(pos_next * 6) + 1];
- float v2n = rb[(pos_next * 6) + 2];
- float v3n = rb[(pos_next * 6) + 3];
- float v4n = rb[(pos_next * 6) + 4];
- float v5n = rb[(pos_next * 6) + 5];
- */
+ v0 += (v0n - v0) * frac;
+ v1 += (v1n - v1) * frac;
p_dest[i] = AudioFrame(v0, v1);
}
}
diff --git a/servers/audio/audio_stream.cpp b/servers/audio/audio_stream.cpp
index 7de0695e8c..02a0bed964 100644
--- a/servers/audio/audio_stream.cpp
+++ b/servers/audio/audio_stream.cpp
@@ -136,16 +136,20 @@ void AudioStreamPlaybackMicrophone::_mix_internal(AudioFrame *p_buffer, int p_fr
Vector<int32_t> buf = AudioDriver::get_singleton()->get_input_buffer();
unsigned int input_size = AudioDriver::get_singleton()->get_input_size();
+ int mix_rate = AudioDriver::get_singleton()->get_mix_rate();
+ int playback_delay = MIN(((50 * mix_rate) / 1000) * 2, buf.size() >> 1);
+#ifdef DEBUG_ENABLED
+ unsigned int input_position = AudioDriver::get_singleton()->get_input_position();
+#endif
- // p_frames is multiplied by two since an AudioFrame is stereo
- if ((p_frames + MICROPHONE_PLAYBACK_DELAY * 2) > input_size) {
+ if (playback_delay > input_size) {
for (int i = 0; i < p_frames; i++) {
p_buffer[i] = AudioFrame(0.0f, 0.0f);
}
input_ofs = 0;
} else {
for (int i = 0; i < p_frames; i++) {
- if (input_size >= input_ofs) {
+ if (input_size > input_ofs) {
float l = (buf[input_ofs++] >> 16) / 32768.f;
if (input_ofs >= buf.size()) {
input_ofs = 0;
@@ -162,6 +166,12 @@ void AudioStreamPlaybackMicrophone::_mix_internal(AudioFrame *p_buffer, int p_fr
}
}
+#ifdef DEBUG_ENABLED
+ if (input_ofs > input_position && (input_ofs - input_position) < (p_frames * 2)) {
+ print_verbose(String(get_class_name()) + " buffer underrun: input_position=" + itos(input_position) + " input_ofs=" + itos(input_ofs) + " input_size=" + itos(input_size));
+ }
+#endif
+
AudioDriver::get_singleton()->unlock();
}
diff --git a/servers/audio/audio_stream.h b/servers/audio/audio_stream.h
index 2740f86d55..f6ed45cc9c 100644
--- a/servers/audio/audio_stream.h
+++ b/servers/audio/audio_stream.h
@@ -122,8 +122,6 @@ class AudioStreamPlaybackMicrophone : public AudioStreamPlaybackResampled {
GDCLASS(AudioStreamPlaybackMicrophone, AudioStreamPlayback)
friend class AudioStreamMicrophone;
- static const int MICROPHONE_PLAYBACK_DELAY = 256;
-
bool active;
unsigned int input_ofs;
diff --git a/servers/audio/effects/audio_effect_filter.h b/servers/audio/effects/audio_effect_filter.h
index 11978882bc..e8f12e5efa 100644
--- a/servers/audio/effects/audio_effect_filter.h
+++ b/servers/audio/effects/audio_effect_filter.h
@@ -96,6 +96,11 @@ VARIANT_ENUM_CAST(AudioEffectFilter::FilterDB)
class AudioEffectLowPassFilter : public AudioEffectFilter {
GDCLASS(AudioEffectLowPassFilter, AudioEffectFilter)
+
+ void _validate_property(PropertyInfo &property) const {
+ if (property.name == "gain") property.usage = 0;
+ }
+
public:
AudioEffectLowPassFilter() :
AudioEffectFilter(AudioFilterSW::LOWPASS) {}
@@ -103,6 +108,10 @@ public:
class AudioEffectHighPassFilter : public AudioEffectFilter {
GDCLASS(AudioEffectHighPassFilter, AudioEffectFilter)
+ void _validate_property(PropertyInfo &property) const {
+ if (property.name == "gain") property.usage = 0;
+ }
+
public:
AudioEffectHighPassFilter() :
AudioEffectFilter(AudioFilterSW::HIGHPASS) {}
@@ -110,6 +119,10 @@ public:
class AudioEffectBandPassFilter : public AudioEffectFilter {
GDCLASS(AudioEffectBandPassFilter, AudioEffectFilter)
+ void _validate_property(PropertyInfo &property) const {
+ if (property.name == "gain") property.usage = 0;
+ }
+
public:
AudioEffectBandPassFilter() :
AudioEffectFilter(AudioFilterSW::BANDPASS) {}
diff --git a/servers/audio/effects/reverb.cpp b/servers/audio/effects/reverb.cpp
index ef23e4aaf3..b032c91da2 100644
--- a/servers/audio/effects/reverb.cpp
+++ b/servers/audio/effects/reverb.cpp
@@ -36,22 +36,22 @@
const float Reverb::comb_tunings[MAX_COMBS] = {
//freeverb comb tunings
- 0.025306122448979593,
- 0.026938775510204082,
- 0.028956916099773241,
- 0.03074829931972789,
- 0.032244897959183672,
- 0.03380952380952381,
- 0.035306122448979592,
- 0.036666666666666667
+ 0.025306122448979593f,
+ 0.026938775510204082f,
+ 0.028956916099773241f,
+ 0.03074829931972789f,
+ 0.032244897959183672f,
+ 0.03380952380952381f,
+ 0.035306122448979592f,
+ 0.036666666666666667f
};
const float Reverb::allpass_tunings[MAX_ALLPASS] = {
//freeverb allpass tunings
- 0.0051020408163265302,
- 0.007732426303854875,
- 0.01,
- 0.012607709750566893
+ 0.0051020408163265302f,
+ 0.007732426303854875f,
+ 0.01f,
+ 0.012607709750566893f
};
void Reverb::process(float *p_src, float *p_dst, int p_frames) {