moonlight-qt/app/streaming/audio/renderers/sdlaud.cpp

#include "sdl.h"

#include <Limelight.h>
#include <SDL.h>

#include <QtGlobal>
#include <QFile>
#include <QTextStream>

SdlAudioRenderer::SdlAudioRenderer()
    : m_AudioDevice(0),
      m_AudioBuffer(nullptr)
{
    SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO));

#ifdef HAVE_MMAL
    // HACK: PulseAudio on Raspberry Pi suffers from horrible underruns,
    // so switch to ALSA if we detect that we're on a Pi. We need to
    // actually set a bogus PULSE_SERVER so it doesn't try to get smart on us
    // and find PA anyway. SDL_AUDIODRIVER=pulseaudio can override this logic.
    if (qgetenv("SDL_AUDIODRIVER").toLower() != "pulseaudio") {
        QFile file("/proc/cpuinfo");
        if (file.open(QIODevice::ReadOnly | QFile::Text)) {
            QTextStream textStream(&file);
            if (textStream.readAll().contains("Raspberry Pi")) {
                SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
                            "Avoiding PulseAudio on Raspberry Pi");

                qputenv("PULSE_SERVER", "");
                if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
                    SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
                                "ALSA failed; falling back to default");
                    qunsetenv("PULSE_SERVER");
                }
                else {
                    SDL_QuitSubSystem(SDL_INIT_AUDIO);
                }
            }
        }
    }
#endif

    if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
                     "SDL_InitSubSystem(SDL_INIT_AUDIO) failed: %s",
                     SDL_GetError());
        SDL_assert(SDL_WasInit(SDL_INIT_AUDIO));
    }
}

bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION* opusConfig)
{
    SDL_AudioSpec want, have;

    SDL_zero(want);
    want.freq = opusConfig->sampleRate;
    want.format = AUDIO_S16;
    want.channels = opusConfig->channelCount;

    // This is supposed to be a power of 2, but our
    // frames contain a non-power of 2 number of samples,
    // so the slop would require buffering another full frame.
    // Specifying non-Po2 seems to work for our supported platforms.
    want.samples = opusConfig->samplesPerFrame;

    m_FrameSize = opusConfig->samplesPerFrame * sizeof(short) * opusConfig->channelCount;

    m_AudioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
    if (m_AudioDevice == 0) {
        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
                     "Failed to open audio device: %s",
                     SDL_GetError());
        return false;
    }

    m_AudioBuffer = malloc(m_FrameSize);
    if (m_AudioBuffer == nullptr) {
        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
                     "Failed to allocate audio buffer");
        return false;
    }

    SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
                "Desired audio buffer: %u samples (%u bytes)",
                want.samples,
                want.samples * (Uint32)sizeof(short) * want.channels);

    SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
                "Obtained audio buffer: %u samples (%u bytes)",
                have.samples,
                have.size);

    // Start playback
    SDL_PauseAudioDevice(m_AudioDevice, 0);

    return true;
}

SdlAudioRenderer::~SdlAudioRenderer()
{
    if (m_AudioDevice != 0) {
        // Stop playback
        SDL_PauseAudioDevice(m_AudioDevice, 1);
        SDL_CloseAudioDevice(m_AudioDevice);
    }

    if (m_AudioBuffer != nullptr) {
        free(m_AudioBuffer);
    }

    SDL_QuitSubSystem(SDL_INIT_AUDIO);
    SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO));
}

void* SdlAudioRenderer::getAudioBuffer(int*)
{
    return m_AudioBuffer;
}

bool SdlAudioRenderer::submitAudio(int bytesWritten)
{
    if (bytesWritten == 0) {
        // Nothing to do
        return true;
    }

    // Don't queue if there's already more than 30 ms of audio data waiting
    // in Moonlight's audio queue.
    if (LiGetPendingAudioDuration() > 30) {
        return true;
    }

    // Provide backpressure on the queue to ensure too many frames don't build up
    // in SDL's audio queue.
    while (SDL_GetQueuedAudioSize(m_AudioDevice) / m_FrameSize > 10) {
        SDL_Delay(1);
    }

    if (SDL_QueueAudio(m_AudioDevice, m_AudioBuffer, bytesWritten) < 0) {
        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
                     "Failed to queue audio sample: %s",
                     SDL_GetError());
    }

    return true;
}

int SdlAudioRenderer::getCapabilities()
{
    // Direct submit can't be used because we use LiGetPendingAudioDuration()
    return CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
}