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https://github.com/dolphin-emu/dolphin.git
synced 2025-07-24 22:59:47 -06:00
Reformat all the things. Have fun with merge conflicts.
This commit is contained in:
Pierre Bourdon
@ -6,230 +6,229 @@
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#include "AudioCommon/AlsaSoundStream.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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AlsaSound::AlsaSound()
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: m_thread_status(ALSAThreadStatus::STOPPED)
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, handle(nullptr)
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, frames_to_deliver(FRAME_COUNT_MIN)
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: m_thread_status(ALSAThreadStatus::STOPPED), handle(nullptr),
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frames_to_deliver(FRAME_COUNT_MIN)
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{
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}
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bool AlsaSound::Start()
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{
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m_thread_status.store(ALSAThreadStatus::RUNNING);
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if (!AlsaInit())
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{
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m_thread_status.store(ALSAThreadStatus::STOPPED);
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return false;
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}
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m_thread_status.store(ALSAThreadStatus::RUNNING);
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if (!AlsaInit())
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{
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m_thread_status.store(ALSAThreadStatus::STOPPED);
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return false;
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}
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thread = std::thread(&AlsaSound::SoundLoop, this);
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return true;
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thread = std::thread(&AlsaSound::SoundLoop, this);
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return true;
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}
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void AlsaSound::Stop()
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{
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m_thread_status.store(ALSAThreadStatus::STOPPING);
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m_thread_status.store(ALSAThreadStatus::STOPPING);
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//Give the opportunity to the audio thread
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//to realize we are stopping the emulation
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cv.notify_one();
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thread.join();
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// Give the opportunity to the audio thread
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// to realize we are stopping the emulation
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cv.notify_one();
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thread.join();
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}
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void AlsaSound::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 AlsaSound::SoundLoop()
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{
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Common::SetCurrentThreadName("Audio thread - alsa");
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while (m_thread_status.load() != ALSAThreadStatus::STOPPING)
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{
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while (m_thread_status.load() == ALSAThreadStatus::RUNNING)
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{
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m_mixer->Mix(mix_buffer, frames_to_deliver);
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int rc = snd_pcm_writei(handle, mix_buffer, frames_to_deliver);
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if (rc == -EPIPE)
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{
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// Underrun
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snd_pcm_prepare(handle);
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}
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else if (rc < 0)
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{
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ERROR_LOG(AUDIO, "writei fail: %s", snd_strerror(rc));
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}
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}
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if (m_thread_status.load() == ALSAThreadStatus::PAUSED)
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{
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snd_pcm_drop(handle); // Stop sound output
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Common::SetCurrentThreadName("Audio thread - alsa");
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while (m_thread_status.load() != ALSAThreadStatus::STOPPING)
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{
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while (m_thread_status.load() == ALSAThreadStatus::RUNNING)
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{
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m_mixer->Mix(mix_buffer, frames_to_deliver);
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int rc = snd_pcm_writei(handle, mix_buffer, frames_to_deliver);
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if (rc == -EPIPE)
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{
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// Underrun
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snd_pcm_prepare(handle);
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}
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else if (rc < 0)
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{
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ERROR_LOG(AUDIO, "writei fail: %s", snd_strerror(rc));
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}
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}
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if (m_thread_status.load() == ALSAThreadStatus::PAUSED)
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{
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snd_pcm_drop(handle); // Stop sound output
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// Block until thread status changes.
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std::unique_lock<std::mutex> lock(cv_m);
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cv.wait(lock, [this]{ return m_thread_status.load() != ALSAThreadStatus::PAUSED; });
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// Block until thread status changes.
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std::unique_lock<std::mutex> lock(cv_m);
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cv.wait(lock, [this] { return m_thread_status.load() != ALSAThreadStatus::PAUSED; });
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snd_pcm_prepare(handle); // resume sound output
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}
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}
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AlsaShutdown();
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m_thread_status.store(ALSAThreadStatus::STOPPED);
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snd_pcm_prepare(handle); // resume sound output
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}
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}
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AlsaShutdown();
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m_thread_status.store(ALSAThreadStatus::STOPPED);
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}
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void AlsaSound::Clear(bool muted)
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{
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m_muted = muted;
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m_thread_status.store(muted ? ALSAThreadStatus::PAUSED : ALSAThreadStatus::RUNNING);
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cv.notify_one(); // Notify thread that status has changed
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m_muted = muted;
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m_thread_status.store(muted ? ALSAThreadStatus::PAUSED : ALSAThreadStatus::RUNNING);
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cv.notify_one(); // Notify thread that status has changed
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}
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bool AlsaSound::AlsaInit()
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{
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unsigned int sample_rate = m_mixer->GetSampleRate();
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int err;
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int dir;
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snd_pcm_sw_params_t *swparams;
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snd_pcm_hw_params_t *hwparams;
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snd_pcm_uframes_t buffer_size,buffer_size_max;
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unsigned int periods;
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unsigned int sample_rate = m_mixer->GetSampleRate();
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int err;
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int dir;
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snd_pcm_sw_params_t* swparams;
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snd_pcm_hw_params_t* hwparams;
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snd_pcm_uframes_t buffer_size, buffer_size_max;
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unsigned int periods;
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err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Audio open error: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Audio open error: %s\n", snd_strerror(err));
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return false;
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}
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snd_pcm_hw_params_alloca(&hwparams);
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snd_pcm_hw_params_alloca(&hwparams);
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err = snd_pcm_hw_params_any(handle, hwparams);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Broken configuration for this PCM: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_any(handle, hwparams);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Broken configuration for this PCM: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Access type not available: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Access type not available: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Sample format not available: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Sample format not available: %s\n", snd_strerror(err));
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return false;
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}
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dir = 0;
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err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &sample_rate, &dir);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Rate not available: %s\n", snd_strerror(err));
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return false;
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}
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dir = 0;
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err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &sample_rate, &dir);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Rate not available: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_set_channels(handle, hwparams, CHANNEL_COUNT);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Channels count not available: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_set_channels(handle, hwparams, CHANNEL_COUNT);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Channels count not available: %s\n", snd_strerror(err));
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return false;
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}
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periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
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err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Cannot set maximum periods per buffer: %s\n", snd_strerror(err));
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return false;
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}
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periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
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err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Cannot set maximum periods per buffer: %s\n", snd_strerror(err));
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return false;
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}
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buffer_size_max = BUFFER_SIZE_MAX;
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err = snd_pcm_hw_params_set_buffer_size_max(handle, hwparams, &buffer_size_max);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Cannot set maximum buffer size: %s\n", snd_strerror(err));
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return false;
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}
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buffer_size_max = BUFFER_SIZE_MAX;
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err = snd_pcm_hw_params_set_buffer_size_max(handle, hwparams, &buffer_size_max);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Cannot set maximum buffer size: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params(handle, hwparams);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params(handle, hwparams);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Cannot get buffer size: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Cannot get buffer size: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_get_periods_max(hwparams, &periods, &dir);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Cannot get periods: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_hw_params_get_periods_max(hwparams, &periods, &dir);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Cannot get periods: %s\n", snd_strerror(err));
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return false;
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}
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//periods is the number of fragments alsa can wait for during one
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//buffer_size
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frames_to_deliver = buffer_size / periods;
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//limit the minimum size. pulseaudio advertises a minimum of 32 samples.
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if (frames_to_deliver < FRAME_COUNT_MIN)
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frames_to_deliver = FRAME_COUNT_MIN;
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//it is probably a bad idea to try to send more than one buffer of data
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if ((unsigned int)frames_to_deliver > buffer_size)
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frames_to_deliver = buffer_size;
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NOTICE_LOG(AUDIO, "ALSA gave us a %ld sample \"hardware\" buffer with %d periods. Will send %d samples per fragments.\n", buffer_size, periods, frames_to_deliver);
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// periods is the number of fragments alsa can wait for during one
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// buffer_size
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frames_to_deliver = buffer_size / periods;
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// limit the minimum size. pulseaudio advertises a minimum of 32 samples.
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if (frames_to_deliver < FRAME_COUNT_MIN)
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frames_to_deliver = FRAME_COUNT_MIN;
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// it is probably a bad idea to try to send more than one buffer of data
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if ((unsigned int)frames_to_deliver > buffer_size)
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frames_to_deliver = buffer_size;
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NOTICE_LOG(AUDIO, "ALSA gave us a %ld sample \"hardware\" buffer with %d periods. Will send %d "
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"samples per fragments.\n",
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buffer_size, periods, frames_to_deliver);
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snd_pcm_sw_params_alloca(&swparams);
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snd_pcm_sw_params_alloca(&swparams);
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err = snd_pcm_sw_params_current(handle, swparams);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "cannot init sw params: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_sw_params_current(handle, swparams);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "cannot init sw params: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "cannot set start thresh: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "cannot set start thresh: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_sw_params(handle, swparams);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_sw_params(handle, swparams);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
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return false;
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}
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err = snd_pcm_prepare(handle);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Unable to prepare: %s\n", snd_strerror(err));
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return false;
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}
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NOTICE_LOG(AUDIO, "ALSA successfully initialized.\n");
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return true;
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err = snd_pcm_prepare(handle);
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if (err < 0)
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{
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ERROR_LOG(AUDIO, "Unable to prepare: %s\n", snd_strerror(err));
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return false;
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}
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NOTICE_LOG(AUDIO, "ALSA successfully initialized.\n");
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return true;
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}
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void AlsaSound::AlsaShutdown()
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{
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if (handle != nullptr)
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{
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snd_pcm_drop(handle);
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snd_pcm_close(handle);
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handle = nullptr;
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}
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if (handle != nullptr)
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{
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snd_pcm_drop(handle);
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snd_pcm_close(handle);
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handle = nullptr;
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}
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}
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