From 95c912310d9ba9c15e0ec47ab73e5a3614f6ac82 Mon Sep 17 00:00:00 2001 From: Adam Moss Date: Wed, 7 Jan 2015 10:57:20 +0000 Subject: [PATCH] PulseAudio: Support 5.1 surround, optional in UI --- Source/Core/AudioCommon/PulseAudioStream.cpp | 86 +++++++++++++++++--- Source/Core/AudioCommon/PulseAudioStream.h | 4 + Source/Core/DolphinWX/ConfigMain.cpp | 8 +- 3 files changed, 84 insertions(+), 14 deletions(-) diff --git a/Source/Core/AudioCommon/PulseAudioStream.cpp b/Source/Core/AudioCommon/PulseAudioStream.cpp index 8d56e8a377..e474fdf14d 100644 --- a/Source/Core/AudioCommon/PulseAudioStream.cpp +++ b/Source/Core/AudioCommon/PulseAudioStream.cpp @@ -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). diff --git a/Source/Core/AudioCommon/PulseAudioStream.h b/Source/Core/AudioCommon/PulseAudioStream.h index e84bed6462..d5f9d06c20 100644 --- a/Source/Core/AudioCommon/PulseAudioStream.h +++ b/Source/Core/AudioCommon/PulseAudioStream.h @@ -45,6 +45,10 @@ private: std::thread m_thread; std::atomic 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; diff --git a/Source/Core/DolphinWX/ConfigMain.cpp b/Source/Core/DolphinWX/ConfigMain.cpp index 003099a67b..ca45ce9838 100644 --- a/Source/Core/DolphinWX/ConfigMain.cpp +++ b/Source/Core/DolphinWX/ConfigMain.cpp @@ -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;