mirror of
https://github.com/dolphin-emu/dolphin.git
synced 2025-07-21 05:09:34 -06:00
Reformat all the things. Have fun with merge conflicts.
This commit is contained in:
Pierre Bourdon
@ -7,245 +7,247 @@
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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/Thread.h"
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#include "Common/Logging/Log.h"
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#include "Common/Thread.h"
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#include "Core/ConfigManager.h"
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namespace
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{
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const size_t BUFFER_SAMPLES = 512; // ~10 ms - needs to be at least 240 for surround
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const size_t BUFFER_SAMPLES = 512; // ~10 ms - needs to be at least 240 for surround
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}
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PulseAudio::PulseAudio()
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: m_thread()
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, m_run_thread()
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PulseAudio::PulseAudio() : m_thread(), m_run_thread()
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{
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}
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bool PulseAudio::Start()
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{
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m_stereo = !SConfig::GetInstance().bDPL2Decoder;
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m_channels = m_stereo ? 2 : 5; // will tell PA we use a Stereo or 5.0 channel setup
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m_stereo = !SConfig::GetInstance().bDPL2Decoder;
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m_channels = m_stereo ? 2 : 5; // will tell PA we use a Stereo or 5.0 channel setup
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NOTICE_LOG(AUDIO, "PulseAudio backend using %d channels", m_channels);
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NOTICE_LOG(AUDIO, "PulseAudio backend using %d channels", m_channels);
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m_run_thread = true;
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m_thread = std::thread(&PulseAudio::SoundLoop, this);
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m_run_thread = true;
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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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// Initialize DPL2 parameters
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DPL2Reset();
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return true;
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return true;
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}
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void PulseAudio::Stop()
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{
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m_run_thread = false;
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m_thread.join();
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m_run_thread = false;
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m_thread.join();
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}
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void PulseAudio::Update()
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{
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// don't need to do anything here.
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// don't need to do anything here.
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}
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// Called on audio thread.
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void PulseAudio::SoundLoop()
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{
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Common::SetCurrentThreadName("Audio thread - pulse");
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Common::SetCurrentThreadName("Audio thread - pulse");
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if (PulseInit())
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{
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while (m_run_thread.load() && m_pa_connected == 1 && m_pa_error >= 0)
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m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
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if (PulseInit())
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{
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while (m_run_thread.load() && m_pa_connected == 1 && m_pa_error >= 0)
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m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
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if (m_pa_error < 0)
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ERROR_LOG(AUDIO, "PulseAudio error: %s", pa_strerror(m_pa_error));
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if (m_pa_error < 0)
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ERROR_LOG(AUDIO, "PulseAudio error: %s", pa_strerror(m_pa_error));
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PulseShutdown();
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}
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PulseShutdown();
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}
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}
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bool PulseAudio::PulseInit()
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{
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m_pa_error = 0;
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m_pa_connected = 0;
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m_pa_error = 0;
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m_pa_connected = 0;
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// create pulseaudio main loop and context
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// also register the async state callback which is called when the connection to the pa server has changed
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m_pa_ml = pa_mainloop_new();
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m_pa_mlapi = pa_mainloop_get_api(m_pa_ml);
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m_pa_ctx = pa_context_new(m_pa_mlapi, "dolphin-emu");
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m_pa_error = pa_context_connect(m_pa_ctx, nullptr, PA_CONTEXT_NOFLAGS, nullptr);
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pa_context_set_state_callback(m_pa_ctx, StateCallback, this);
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// create pulseaudio main loop and context
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// also register the async state callback which is called when the connection to the pa server has
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// changed
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m_pa_ml = pa_mainloop_new();
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m_pa_mlapi = pa_mainloop_get_api(m_pa_ml);
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m_pa_ctx = pa_context_new(m_pa_mlapi, "dolphin-emu");
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m_pa_error = pa_context_connect(m_pa_ctx, nullptr, PA_CONTEXT_NOFLAGS, nullptr);
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pa_context_set_state_callback(m_pa_ctx, StateCallback, this);
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// wait until we're connected to the pulseaudio server
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while (m_pa_connected == 0 && m_pa_error >= 0)
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m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
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// wait until we're connected to the pulseaudio server
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while (m_pa_connected == 0 && m_pa_error >= 0)
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m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
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if (m_pa_connected == 2 || m_pa_error < 0)
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{
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ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
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return false;
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}
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if (m_pa_connected == 2 || m_pa_error < 0)
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{
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ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
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return false;
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}
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// create a new audio stream with our sample format
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// also connect the callbacks for this stream
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pa_sample_spec ss;
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pa_channel_map channel_map;
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pa_channel_map* channel_map_p = nullptr; // auto channel map
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if (m_stereo)
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{
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ss.format = PA_SAMPLE_S16LE;
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m_bytespersample = sizeof(s16);
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}
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else
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{
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// surround is remixed in floats, use a float PA buffer to save another conversion
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ss.format = PA_SAMPLE_FLOAT32NE;
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m_bytespersample = sizeof(float);
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// create a new audio stream with our sample format
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// also connect the callbacks for this stream
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pa_sample_spec ss;
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pa_channel_map channel_map;
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pa_channel_map* channel_map_p = nullptr; // auto channel map
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if (m_stereo)
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{
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ss.format = PA_SAMPLE_S16LE;
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m_bytespersample = sizeof(s16);
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}
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else
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{
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// surround is remixed in floats, use a float PA buffer to save another conversion
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ss.format = PA_SAMPLE_FLOAT32NE;
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m_bytespersample = sizeof(float);
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channel_map_p = &channel_map; // explicit channel map:
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channel_map.channels = 5;
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channel_map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
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channel_map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
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channel_map.map[2] = PA_CHANNEL_POSITION_FRONT_CENTER;
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channel_map.map[3] = PA_CHANNEL_POSITION_REAR_LEFT;
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channel_map.map[4] = PA_CHANNEL_POSITION_REAR_RIGHT;
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}
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ss.channels = m_channels;
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ss.rate = m_mixer->GetSampleRate();
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assert(pa_sample_spec_valid(&ss));
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m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, channel_map_p);
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pa_stream_set_write_callback(m_pa_s, WriteCallback, this);
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pa_stream_set_underflow_callback(m_pa_s, UnderflowCallback, this);
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channel_map_p = &channel_map; // explicit channel map:
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channel_map.channels = 5;
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channel_map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
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channel_map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
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channel_map.map[2] = PA_CHANNEL_POSITION_FRONT_CENTER;
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channel_map.map[3] = PA_CHANNEL_POSITION_REAR_LEFT;
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channel_map.map[4] = PA_CHANNEL_POSITION_REAR_RIGHT;
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}
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ss.channels = m_channels;
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ss.rate = m_mixer->GetSampleRate();
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assert(pa_sample_spec_valid(&ss));
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m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, channel_map_p);
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pa_stream_set_write_callback(m_pa_s, WriteCallback, this);
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pa_stream_set_underflow_callback(m_pa_s, UnderflowCallback, this);
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// connect this audio stream to the default audio playback
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// limit buffersize to reduce latency
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m_pa_ba.fragsize = -1;
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m_pa_ba.maxlength = -1; // max buffer, so also max latency
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m_pa_ba.minreq = -1; // don't read every byte, try to group them _a bit_
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m_pa_ba.prebuf = -1; // start as early as possible
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m_pa_ba.tlength = BUFFER_SAMPLES * m_channels * m_bytespersample; // designed latency, only change this flag for low latency output
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pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
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m_pa_error = pa_stream_connect_playback(m_pa_s, nullptr, &m_pa_ba, flags, nullptr, nullptr);
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if (m_pa_error < 0)
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{
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ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
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return false;
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}
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// connect this audio stream to the default audio playback
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// limit buffersize to reduce latency
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m_pa_ba.fragsize = -1;
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m_pa_ba.maxlength = -1; // max buffer, so also max latency
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m_pa_ba.minreq = -1; // don't read every byte, try to group them _a bit_
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m_pa_ba.prebuf = -1; // start as early as possible
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m_pa_ba.tlength =
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BUFFER_SAMPLES * m_channels *
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m_bytespersample; // designed latency, only change this flag for low latency output
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pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY |
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PA_STREAM_AUTO_TIMING_UPDATE);
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m_pa_error = pa_stream_connect_playback(m_pa_s, nullptr, &m_pa_ba, flags, nullptr, nullptr);
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if (m_pa_error < 0)
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{
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ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
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return false;
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}
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INFO_LOG(AUDIO, "Pulse successfully initialized");
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return true;
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INFO_LOG(AUDIO, "Pulse successfully initialized");
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return true;
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}
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void PulseAudio::PulseShutdown()
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{
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pa_context_disconnect(m_pa_ctx);
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pa_context_unref(m_pa_ctx);
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pa_mainloop_free(m_pa_ml);
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pa_context_disconnect(m_pa_ctx);
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pa_context_unref(m_pa_ctx);
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pa_mainloop_free(m_pa_ml);
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}
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void PulseAudio::StateCallback(pa_context* c)
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{
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pa_context_state_t state = pa_context_get_state(c);
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switch (state)
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{
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case PA_CONTEXT_FAILED:
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case PA_CONTEXT_TERMINATED:
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m_pa_connected = 2;
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break;
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case PA_CONTEXT_READY:
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m_pa_connected = 1;
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break;
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default:
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break;
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}
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pa_context_state_t state = pa_context_get_state(c);
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switch (state)
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{
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case PA_CONTEXT_FAILED:
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case PA_CONTEXT_TERMINATED:
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m_pa_connected = 2;
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break;
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case PA_CONTEXT_READY:
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m_pa_connected = 1;
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break;
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default:
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break;
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}
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}
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// on underflow, increase pulseaudio latency in ~10ms steps
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void PulseAudio::UnderflowCallback(pa_stream* s)
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{
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m_pa_ba.tlength += BUFFER_SAMPLES * m_channels * m_bytespersample;
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pa_operation* op = pa_stream_set_buffer_attr(s, &m_pa_ba, nullptr, nullptr);
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pa_operation_unref(op);
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m_pa_ba.tlength += BUFFER_SAMPLES * m_channels * m_bytespersample;
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pa_operation* op = pa_stream_set_buffer_attr(s, &m_pa_ba, nullptr, nullptr);
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pa_operation_unref(op);
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WARN_LOG(AUDIO, "pulseaudio underflow, new latency: %d bytes", m_pa_ba.tlength);
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WARN_LOG(AUDIO, "pulseaudio underflow, new latency: %d bytes", m_pa_ba.tlength);
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}
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void PulseAudio::WriteCallback(pa_stream* s, size_t length)
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{
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int bytes_per_frame = m_channels * m_bytespersample;
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int frames = (length / bytes_per_frame);
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size_t trunc_length = frames * bytes_per_frame;
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int bytes_per_frame = m_channels * m_bytespersample;
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int frames = (length / bytes_per_frame);
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size_t trunc_length = frames * bytes_per_frame;
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// fetch dst buffer directly from pulseaudio, so no memcpy is needed
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void* buffer;
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m_pa_error = pa_stream_begin_write(s, &buffer, &trunc_length);
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// fetch dst buffer directly from pulseaudio, so no memcpy is needed
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void* buffer;
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m_pa_error = pa_stream_begin_write(s, &buffer, &trunc_length);
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if (!buffer || m_pa_error < 0)
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return; // error will be printed from main loop
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if (!buffer || m_pa_error < 0)
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return; // error will be printed from main loop
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if (m_stereo)
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{
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// use the raw s16 stereo mix
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m_mixer->Mix((s16*) buffer, frames);
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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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if (m_stereo)
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{
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// use the raw s16 stereo mix
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m_mixer->Mix((s16*)buffer, frames);
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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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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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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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// 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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for (int i=0; i < frames; ++i)
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{
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for (int j=0; j < m_channels; ++j)
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{
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((float*)buffer)[m_channels * i + j] = floatbuffer_6chan[6 * i + dpl2_to_5chan[j]];
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}
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}
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}
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else
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{
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ERROR_LOG(AUDIO, "Unsupported number of PA channels requested: %d", (int)m_channels);
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return;
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}
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}
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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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for (int i = 0; i < frames; ++i)
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{
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for (int j = 0; j < m_channels; ++j)
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{
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((float*)buffer)[m_channels * i + j] = floatbuffer_6chan[6 * i + dpl2_to_5chan[j]];
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}
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}
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}
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else
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{
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ERROR_LOG(AUDIO, "Unsupported number of PA channels requested: %d", (int)m_channels);
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return;
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}
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}
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m_pa_error = pa_stream_write(s, buffer, trunc_length, nullptr, 0, PA_SEEK_RELATIVE);
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m_pa_error = pa_stream_write(s, buffer, trunc_length, nullptr, 0, PA_SEEK_RELATIVE);
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}
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// Callbacks that forward to internal methods (required because PulseAudio is a C API).
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void PulseAudio::StateCallback(pa_context* c, void* userdata)
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{
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PulseAudio* p = (PulseAudio*) userdata;
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p->StateCallback(c);
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PulseAudio* p = (PulseAudio*)userdata;
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p->StateCallback(c);
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}
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void PulseAudio::UnderflowCallback(pa_stream* s, void* userdata)
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{
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PulseAudio* p = (PulseAudio*) userdata;
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p->UnderflowCallback(s);
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PulseAudio* p = (PulseAudio*)userdata;
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p->UnderflowCallback(s);
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}
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void PulseAudio::WriteCallback(pa_stream* s, size_t length, void* userdata)
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{
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PulseAudio* p = (PulseAudio*) userdata;
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p->WriteCallback(s, length);
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PulseAudio* p = (PulseAudio*)userdata;
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p->WriteCallback(s, length);
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}
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