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