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PulseAudio: Support 5.1 surround, optional in UI

This commit is contained in:
Adam Moss 2015-01-07 10:57:20 +00:00 committed by Adam D. Moss
parent 25fac0282c
commit 95c912310d
3 changed files with 84 additions and 14 deletions
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86
Source/Core/AudioCommon/PulseAudioStream.cpp
View File

@ -2,15 +2,15 @@
// Licensed under GPLv2
// Refer to the license.txt file included.
#include "AudioCommon/DPL2Decoder.h"
#include "AudioCommon/PulseAudioStream.h"
#include "Common/CommonTypes.h"
#include "Common/Thread.h"
#include "Core/ConfigManager.h"
namespace
{
const size_t BUFFER_SAMPLES = 512; // ~10 ms
const size_t CHANNEL_COUNT = 2;
const size_t BUFFER_SIZE = BUFFER_SAMPLES * CHANNEL_COUNT * sizeof(s16);
const size_t BUFFER_SAMPLES = 512; // ~10 ms - needs to be at least 240 for surround
}
PulseAudio::PulseAudio(CMixer *mixer)
@ -22,8 +22,17 @@ PulseAudio::PulseAudio(CMixer *mixer)
bool PulseAudio::Start()
{
m_stereo = !SConfig::GetInstance().m_LocalCoreStartupParameter.bDPL2Decoder;
m_channels = m_stereo ? 2 : 6; // will tell PA we use a Stereo or 5.1 channel setup
NOTICE_LOG(AUDIO, "PulseAudio backend using %d channels", m_channels);
m_run_thread = true;
m_thread = std::thread(&PulseAudio::SoundLoop, this);
// Initialize DPL2 parameters
DPL2Reset();
return true;
}
@ -81,10 +90,33 @@ bool PulseAudio::PulseInit()
// create a new audio stream with our sample format
// also connect the callbacks for this stream
pa_sample_spec ss;
ss.format = PA_SAMPLE_S16LE;
ss.channels = 2;
pa_channel_map channel_map;
pa_channel_map* channel_map_p = nullptr; // auto channel map
if (m_stereo)
{
ss.format = PA_SAMPLE_S16LE;
m_bytespersample = sizeof(s16);
}
else
{
// surround is remixed in floats, use a float PA buffer to save another conversion
ss.format = PA_SAMPLE_FLOAT32NE;
m_bytespersample = sizeof(float);
// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
channel_map_p = &channel_map; // explicit channel map:
channel_map.channels = 6;
channel_map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
channel_map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
channel_map.map[2] = PA_CHANNEL_POSITION_FRONT_CENTER;
channel_map.map[3] = PA_CHANNEL_POSITION_LFE;
channel_map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT;
channel_map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT;
}
ss.channels = m_channels;
ss.rate = m_mixer->GetSampleRate();
m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, nullptr);
assert(pa_sample_spec_valid(&ss));
m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, channel_map_p);
pa_stream_set_write_callback(m_pa_s, WriteCallback, this);
pa_stream_set_underflow_callback(m_pa_s, UnderflowCallback, this);
@ -94,7 +126,7 @@ bool PulseAudio::PulseInit()
m_pa_ba.maxlength = -1; // max buffer, so also max latency
m_pa_ba.minreq = -1; // don't read every byte, try to group them _a bit_
m_pa_ba.prebuf = -1; // start as early as possible
m_pa_ba.tlength = BUFFER_SIZE; // designed latency, only change this flag for low latency output
m_pa_ba.tlength = BUFFER_SAMPLES * m_channels * m_bytespersample; // designed latency, only change this flag for low latency output
pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
m_pa_error = pa_stream_connect_playback(m_pa_s, nullptr, &m_pa_ba, flags, nullptr, nullptr);
if (m_pa_error < 0)
@ -133,7 +165,7 @@ void PulseAudio::StateCallback(pa_context* c)
// on underflow, increase pulseaudio latency in ~10ms steps
void PulseAudio::UnderflowCallback(pa_stream* s)
{
m_pa_ba.tlength += BUFFER_SIZE;
m_pa_ba.tlength += BUFFER_SAMPLES * m_channels * m_bytespersample;
pa_stream_set_buffer_attr(s, &m_pa_ba, nullptr, nullptr);
WARN_LOG(AUDIO, "pulseaudio underflow, new latency: %d bytes", m_pa_ba.tlength);
@ -141,15 +173,47 @@ void PulseAudio::UnderflowCallback(pa_stream* s)
void PulseAudio::WriteCallback(pa_stream* s, size_t length)
{
int bytes_per_frame = m_channels * m_bytespersample;
int frames = (length / bytes_per_frame);
size_t trunc_length = frames * bytes_per_frame;
// fetch dst buffer directly from pulseaudio, so no memcpy is needed
void* buffer;
m_pa_error = pa_stream_begin_write(s, &buffer, &length);
m_pa_error = pa_stream_begin_write(s, &buffer, &trunc_length);
if (!buffer || m_pa_error < 0)
return; // error will be printed from main loop
m_mixer->Mix((s16*) buffer, length / sizeof(s16) / CHANNEL_COUNT);
m_pa_error = pa_stream_write(s, buffer, length, nullptr, 0, PA_SEEK_RELATIVE);
if (m_stereo)
{
// use the raw s16 stereo mix
m_mixer->Mix((s16*) buffer, frames);
}
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 == 6) // Extract dpl2/5.1 Surround
{
DPL2Decode(floatbuffer_stereo, frames, (float*)buffer);
}
else
{
ERROR_LOG(AUDIO, "Unsupported number of PA channels requested: %d", (int)m_channels);
return;
}
}
m_pa_error = pa_stream_write(s, buffer, trunc_length, nullptr, 0, PA_SEEK_RELATIVE);
}
// Callbacks that forward to internal methods (required because PulseAudio is a C API).

4
Source/Core/AudioCommon/PulseAudioStream.h
View File

@ -45,6 +45,10 @@ private:
std::thread m_thread;
std::atomic<bool> m_run_thread;
bool m_stereo; // stereo, else surround
int m_bytespersample;
int m_channels;
int m_pa_error;
int m_pa_connected;
pa_mainloop *m_pa_ml;

8
Source/Core/DolphinWX/ConfigMain.cpp
View File

@ -359,7 +359,8 @@ void CConfigMain::InitializeGUIValues()
VolumeSlider->Enable(SupportsVolumeChanges(SConfig::GetInstance().sBackend));
VolumeSlider->SetValue(SConfig::GetInstance().m_Volume);
VolumeText->SetLabel(wxString::Format("%d %%", SConfig::GetInstance().m_Volume));
DPL2Decoder->Enable(std::string(SConfig::GetInstance().sBackend) == BACKEND_OPENAL);
DPL2Decoder->Enable(std::string(SConfig::GetInstance().sBackend) == BACKEND_OPENAL
|| std::string(SConfig::GetInstance().sBackend) == BACKEND_PULSEAUDIO);
DPL2Decoder->SetValue(startup_params.bDPL2Decoder);
Latency->Enable(std::string(SConfig::GetInstance().sBackend) == BACKEND_OPENAL);
Latency->SetValue(startup_params.iLatency);
@ -479,7 +480,7 @@ void CConfigMain::InitializeGUITooltips()
#if defined(__APPLE__)
DPL2Decoder->SetToolTip(_("Enables Dolby Pro Logic II emulation using 5.1 surround. Not available on OSX."));
#else
DPL2Decoder->SetToolTip(_("Enables Dolby Pro Logic II emulation using 5.1 surround. OpenAL backend only."));
DPL2Decoder->SetToolTip(_("Enables Dolby Pro Logic II emulation using 5.1 surround. OpenAL or Pulse backends only."));
#endif
Latency->SetToolTip(_("Sets the latency (in ms). Higher values may reduce audio crackling. OpenAL backend only."));
@ -897,7 +898,8 @@ void CConfigMain::AudioSettingsChanged(wxCommandEvent& event)
case ID_BACKEND:
VolumeSlider->Enable(SupportsVolumeChanges(WxStrToStr(BackendSelection->GetStringSelection())));
Latency->Enable(WxStrToStr(BackendSelection->GetStringSelection()) == BACKEND_OPENAL);
DPL2Decoder->Enable(WxStrToStr(BackendSelection->GetStringSelection()) == BACKEND_OPENAL);
DPL2Decoder->Enable(WxStrToStr(BackendSelection->GetStringSelection()) == BACKEND_OPENAL
|| WxStrToStr(BackendSelection->GetStringSelection()) == BACKEND_PULSEAUDIO);
// Don't save the translated BACKEND_NULLSOUND string
SConfig::GetInstance().sBackend = BackendSelection->GetSelection() ?
WxStrToStr(BackendSelection->GetStringSelection()) : BACKEND_NULLSOUND;