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authorMichael Maltese <michaeljosephmaltese@gmail.com>2017-04-23 18:05:21 -0700
committerMichael Maltese <michaeljosephmaltese@gmail.com>2017-06-03 16:53:59 -0700
commita4508e85e8b63e085eac9dca877fae9f7d65e09a (patch)
tree24dc9c24b1ba4aaf85d0d0ae31fe8b8c05ee6922 /Source
parent0e6bd74ed626dd25a85682b90426c26cb10d6a16 (diff)
AudioCommon: move DPL2 decoding into Mixer
Diffstat (limited to 'Source')
-rw-r--r--Source/Core/AudioCommon/CubebStream.cpp22
-rw-r--r--Source/Core/AudioCommon/Mixer.cpp22
-rw-r--r--Source/Core/AudioCommon/Mixer.h2
-rw-r--r--Source/Core/AudioCommon/OpenALStream.cpp32
-rw-r--r--Source/Core/AudioCommon/PulseAudioStream.cpp19
5 files changed, 43 insertions, 54 deletions
diff --git a/Source/Core/AudioCommon/CubebStream.cpp b/Source/Core/AudioCommon/CubebStream.cpp
index c9052e29c9..272060cd74 100644
--- a/Source/Core/AudioCommon/CubebStream.cpp
+++ b/Source/Core/AudioCommon/CubebStream.cpp
@@ -21,29 +21,9 @@ long CubebStream::DataCallback(cubeb_stream* stream, void* user_data, const void
auto* self = static_cast<CubebStream*>(user_data);
if (self->m_stereo)
- {
self->m_mixer->Mix(static_cast<short*>(output_buffer), num_frames);
- }
else
- {
- size_t required_capacity = num_frames * 2;
- if (required_capacity > self->m_short_buffer.capacity() ||
- required_capacity > self->m_floatstereo_buffer.capacity())
- {
- INFO_LOG(AUDIO, "Expanding conversion buffers size: %li frames", num_frames);
- self->m_short_buffer.reserve(required_capacity);
- self->m_floatstereo_buffer.reserve(required_capacity);
- }
-
- self->m_mixer->Mix(self->m_short_buffer.data(), num_frames);
-
- // s16 to float
- for (size_t i = 0; i < static_cast<size_t>(num_frames) * 2; ++i)
- self->m_floatstereo_buffer[i] = self->m_short_buffer[i] / static_cast<float>(1 << 15);
-
- // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
- DPL2Decode(self->m_floatstereo_buffer.data(), num_frames, static_cast<float*>(output_buffer));
- }
+ self->m_mixer->MixSurround(static_cast<float*>(output_buffer), num_frames);
return num_frames;
}
diff --git a/Source/Core/AudioCommon/Mixer.cpp b/Source/Core/AudioCommon/Mixer.cpp
index 8149da088c..2454042fe2 100644
--- a/Source/Core/AudioCommon/Mixer.cpp
+++ b/Source/Core/AudioCommon/Mixer.cpp
@@ -7,6 +7,7 @@
#include <cmath>
#include <cstring>
+#include "AudioCommon/DPL2Decoder.h"
#include "Common/CommonTypes.h"
#include "Common/Logging/Log.h"
#include "Common/MathUtil.h"
@@ -25,6 +26,8 @@ CMixer::CMixer(unsigned int BackendSampleRate) : m_sampleRate(BackendSampleRate)
m_sound_touch.setSetting(SETTING_SEQUENCE_MS, 62);
m_sound_touch.setSetting(SETTING_SEEKWINDOW_MS, 28);
m_sound_touch.setSetting(SETTING_OVERLAP_MS, 8);
+
+ DPL2Reset();
}
CMixer::~CMixer()
@@ -159,6 +162,25 @@ unsigned int CMixer::Mix(short* samples, unsigned int num_samples)
return num_samples;
}
+unsigned int CMixer::MixSurround(float* samples, unsigned int num_samples)
+{
+ if (!num_samples)
+ return 0;
+
+ memset(samples, 0, num_samples * 6 * sizeof(float));
+
+ unsigned int available_samples = Mix(m_stretch_buffer.data(), num_samples);
+ for (size_t i = 0; i < static_cast<size_t>(available_samples) * 2; ++i)
+ {
+ m_float_conversion_buffer[i] =
+ m_stretch_buffer[i] / static_cast<float>(std::numeric_limits<short>::max());
+ }
+
+ DPL2Decode(m_float_conversion_buffer.data(), available_samples, samples);
+
+ return available_samples;
+}
+
void CMixer::StretchAudio(const short* in, unsigned int num_in, short* out, unsigned int num_out)
{
const double time_delta = static_cast<double>(num_out) / m_sampleRate; // seconds
diff --git a/Source/Core/AudioCommon/Mixer.h b/Source/Core/AudioCommon/Mixer.h
index f2127cceba..3afc39b682 100644
--- a/Source/Core/AudioCommon/Mixer.h
+++ b/Source/Core/AudioCommon/Mixer.h
@@ -21,6 +21,7 @@ public:
// Called from audio threads
unsigned int Mix(short* samples, unsigned int numSamples);
+ unsigned int MixSurround(float* samples, unsigned int num_samples);
// Called from main thread
void PushSamples(const short* samples, unsigned int num_samples);
@@ -87,6 +88,7 @@ private:
double m_stretch_ratio = 1.0;
std::array<short, 2> m_last_stretched_sample = {};
std::array<short, MAX_SAMPLES * 2> m_stretch_buffer;
+ std::array<float, MAX_SAMPLES * 2> m_float_conversion_buffer;
WaveFileWriter m_wave_writer_dtk;
WaveFileWriter m_wave_writer_dsp;
diff --git a/Source/Core/AudioCommon/OpenALStream.cpp b/Source/Core/AudioCommon/OpenALStream.cpp
index 3d4bc90e1d..ca343be005 100644
--- a/Source/Core/AudioCommon/OpenALStream.cpp
+++ b/Source/Core/AudioCommon/OpenALStream.cpp
@@ -6,7 +6,6 @@
#include <cstring>
#include <thread>
-#include "AudioCommon/DPL2Decoder.h"
#include "AudioCommon/OpenALStream.h"
#include "AudioCommon/aldlist.h"
#include "Common/Logging/Log.h"
@@ -66,9 +65,6 @@ bool OpenALStream::Start()
PanicAlertT("OpenAL: can't find sound devices");
}
- // Initialize DPL2 parameters
- DPL2Reset();
-
return bReturn;
}
@@ -228,23 +224,18 @@ void OpenALStream::SoundLoop()
numBuffersQueued -= numBuffersProcessed;
}
- // DPL2 accepts 240 samples minimum (FWRDURATION)
- unsigned int minSamples = surround_capable ? 240 : 0;
-
unsigned int numSamples = OAL_MAX_SAMPLES;
- numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
-
- // Convert the samples from short to float
- for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
- sampleBuffer[i] = static_cast<float>(realtimeBuffer[i]) / (1 << 15);
-
- if (numSamples <= minSamples)
- continue;
if (surround_capable)
{
+ // DPL2 accepts 240 samples minimum (FWRDURATION)
+ unsigned int minSamples = 240;
+
float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
- DPL2Decode(sampleBuffer, numSamples, dpl2);
+ numSamples = m_mixer->MixSurround(dpl2, numSamples);
+
+ if (numSamples < minSamples)
+ continue;
// zero-out the subwoofer channel - DPL2Decode generates a pretty
// good 5.0 but not a good 5.1 output. Sadly there is not a 5.0
@@ -311,6 +302,15 @@ void OpenALStream::SoundLoop()
}
else
{
+ numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
+
+ // Convert the samples from short to float
+ for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
+ sampleBuffer[i] = static_cast<float>(realtimeBuffer[i]) / (1 << 15);
+
+ if (!numSamples)
+ continue;
+
if (float32_capable)
{
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
diff --git a/Source/Core/AudioCommon/PulseAudioStream.cpp b/Source/Core/AudioCommon/PulseAudioStream.cpp
index e6e6a45ce2..827816d4e8 100644
--- a/Source/Core/AudioCommon/PulseAudioStream.cpp
+++ b/Source/Core/AudioCommon/PulseAudioStream.cpp
@@ -4,7 +4,6 @@
#include <cstring>
-#include "AudioCommon/DPL2Decoder.h"
#include "AudioCommon/PulseAudioStream.h"
#include "Common/CommonTypes.h"
#include "Common/Logging/Log.h"
@@ -30,9 +29,6 @@ bool PulseAudio::Start()
m_run_thread.Set();
m_thread = std::thread(&PulseAudio::SoundLoop, this);
- // Initialize DPL2 parameters
- DPL2Reset();
-
return true;
}
@@ -194,23 +190,12 @@ void PulseAudio::WriteCallback(pa_stream* s, size_t length)
}
else
{
- // get a floating point mix
- s16 s16buffer_stereo[frames * 2];
- m_mixer->Mix(s16buffer_stereo, frames); // implicitly mixes to 16-bit stereo
-
- float floatbuffer_stereo[frames * 2];
- // s16 to float
- for (int i = 0; i < frames * 2; ++i)
- {
- floatbuffer_stereo[i] = s16buffer_stereo[i] / float(1 << 15);
- }
-
if (m_channels == 5) // Extract dpl2/5.0 Surround
{
float floatbuffer_6chan[frames * 6];
- // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
- DPL2Decode(floatbuffer_stereo, frames, floatbuffer_6chan);
+ m_mixer->MixSurround(floatbuffer_6chan, frames);
+ // DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
// Discard the subwoofer channel - DPL2Decode generates a pretty
// good 5.0 but not a good 5.1 output.
const int dpl2_to_5chan[] = {0, 1, 2, 4, 5};