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ALSA and AQS use floating-point audio output now
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f4dc3626e5
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befa1a11a2
@ -135,7 +135,13 @@ struct ALSAAudioVoiceEngine : BaseAudioVoiceEngine
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snd_pcm_hw_params_any(m_pcm, hwParams);
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snd_pcm_format_t bestFmt;
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if (!snd_pcm_hw_params_test_format(m_pcm, hwParams, SND_PCM_FORMAT_S32))
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if (!snd_pcm_hw_params_test_format(m_pcm, hwParams, SND_PCM_FORMAT_FLOAT))
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{
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bestFmt = SND_PCM_FORMAT_FLOAT;
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m_mixInfo.m_sampleFormat = SOXR_FLOAT32_I;
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m_mixInfo.m_bitsPerSample = 32;
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}
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else if (!snd_pcm_hw_params_test_format(m_pcm, hwParams, SND_PCM_FORMAT_S32))
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{
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bestFmt = SND_PCM_FORMAT_S32;
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m_mixInfo.m_sampleFormat = SOXR_INT32_I;
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@ -56,7 +56,7 @@ struct AQSAudioVoiceEngine : BaseAudioVoiceEngine
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engine->m_engineEnterCv.wait(lk);
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engine->m_cbWaiting = false;
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engine->_pumpAndMixVoices(engine->m_mixInfo.m_periodFrames, reinterpret_cast<int32_t*>(inBuffer->mAudioData));
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engine->_pumpAndMixVoices(engine->m_mixInfo.m_periodFrames, reinterpret_cast<float*>(inBuffer->mAudioData));
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inBuffer->mAudioDataByteSize = engine->m_frameBytes;
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AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, nullptr);
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@ -71,7 +71,7 @@ struct AQSAudioVoiceEngine : BaseAudioVoiceEngine
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AudioStreamBasicDescription desc = {};
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desc.mSampleRate = 96000;
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desc.mFormatID = kAudioFormatLinearPCM;
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desc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
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desc.mFormatFlags = kLinearPCMFormatFlagIsFloat;
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desc.mBytesPerPacket = chCount * 4;
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desc.mFramesPerPacket = 1;
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desc.mBytesPerFrame = chCount * 4;
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@ -545,7 +545,7 @@ struct AQSAudioVoiceEngine : BaseAudioVoiceEngine
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AudioStreamBasicDescription desc = {};
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desc.mSampleRate = 96000;
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desc.mFormatID = kAudioFormatLinearPCM;
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desc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
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desc.mFormatFlags = kLinearPCMFormatFlagIsFloat;
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desc.mBytesPerPacket = chCount * 4;
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desc.mFramesPerPacket = 1;
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desc.mBytesPerFrame = chCount * 4;
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@ -579,7 +579,7 @@ struct AQSAudioVoiceEngine : BaseAudioVoiceEngine
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}
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m_mixInfo.m_sampleRate = actualSampleRate;
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m_mixInfo.m_sampleFormat = SOXR_INT32_I;
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m_mixInfo.m_sampleFormat = SOXR_FLOAT32_I;
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m_mixInfo.m_bitsPerSample = 32;
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m_5msFrames = actualSampleRate * 5 / 1000;
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@ -676,7 +676,7 @@ struct AQSAudioVoiceEngine : BaseAudioVoiceEngine
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for (unsigned i=0 ; i<3 ; ++i)
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{
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_pumpAndMixVoices(m_mixInfo.m_periodFrames, reinterpret_cast<int32_t*>(m_buffers[i]->mAudioData));
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_pumpAndMixVoices(m_mixInfo.m_periodFrames, reinterpret_cast<float*>(m_buffers[i]->mAudioData));
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m_buffers[i]->mAudioDataByteSize = m_frameBytes;
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AudioQueueEnqueueBuffer(m_queue, m_buffers[i], 0, nullptr);
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}
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@ -127,7 +127,7 @@ void AudioVoiceMono::_resetSampleRate(double sampleRate)
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{
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std::vector<float>& scratchFlt = m_root.m_scratchFlt;
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if (scratchFlt.size() < m_root.m_5msFrames)
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scratchFlt.resize(m_root.m_5msFrames);
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scratchFlt.resize(m_root.m_5msFrames + 2);
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soxr_output(m_src, scratchFlt.data(), m_root.m_5msFrames);
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break;
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}
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@ -196,7 +196,7 @@ size_t AudioVoiceMono::pumpAndMix(const AudioVoiceEngineMixInfo& mixInfo,
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{
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std::vector<float>& scratchFlt = m_root.m_scratchFlt;
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if (scratchFlt.size() < frames)
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scratchFlt.resize(frames);
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scratchFlt.resize(frames + 2);
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size_t oDone = soxr_output(m_src, scratchFlt.data(), frames);
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_midUpdate();
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@ -316,7 +316,7 @@ void AudioVoiceStereo::_resetSampleRate(double sampleRate)
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{
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std::vector<float>& scratchFlt = m_root.m_scratchFlt;
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if (scratchFlt.size() < m_root.m_5msFrames * 2)
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scratchFlt.resize(m_root.m_5msFrames * 2);
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scratchFlt.resize(m_root.m_5msFrames * 2 + 2);
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soxr_output(m_src, scratchFlt.data(), m_root.m_5msFrames);
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break;
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}
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@ -389,7 +389,7 @@ size_t AudioVoiceStereo::pumpAndMix(const AudioVoiceEngineMixInfo& mixInfo,
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std::vector<float>& scratchFlt = m_root.m_scratchFlt;
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size_t samples = frames * 2;
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if (scratchFlt.size() < samples)
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scratchFlt.resize(samples);
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scratchFlt.resize(samples + 4);
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size_t oDone = soxr_output(m_src, scratchFlt.data(), frames);
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_midUpdate();
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