[cmake] refactor: Use CPM over submodules (#143)

Transfers the majority of submodules and large externals to CPM, using source archives rather than full Git clones. Not only does this save massive amounts of clone and configure time, but dependencies are grabbed on-demand rather than being required by default. Additionally, CPM will (generally) automatically search for system dependencies, though certain dependencies have options to control this.

Testing shows gains ranging from 5x to 10x in terms of overall clone/configure time.

Reviewed-on: https://git.eden-emu.dev/eden-emu/eden/pulls/143
Reviewed-by: CamilleLaVey <camillelavey99@gmail.com>

1 files changed, 0 insertions, 82 deletions

diff --git a/externals/oboe/samples/iolib/src/main/cpp/player/OneShotSampleSource.cpp b/externals/oboe/samples/iolib/src/main/cpp/player/OneShotSampleSource.cpp
deleted file mode 100644
index 74fef60bd3..0000000000
--- a/externals/oboe/samples/iolib/src/main/cpp/player/OneShotSampleSource.cpp
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (C) 2019 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <string.h>
-
-#include "wav/WavStreamReader.h"
-
-#include "OneShotSampleSource.h"
-
-namespace iolib {
-
-void OneShotSampleSource::mixAudio(float* outBuff, int numChannels, int32_t numFrames) {
-    int32_t numSamples = mSampleBuffer->getNumSamples();
-    int32_t sampleChannels = mSampleBuffer->getProperties().channelCount;
-    int32_t totalSamplesNeeded = numFrames * numChannels; // Total samples to fill the output buffer
-    int32_t samplesProcessed = 0;
-    bool isLoopMode = mIsLoopMode;
-
-    while (samplesProcessed < totalSamplesNeeded && mIsPlaying) {
-        int32_t samplesLeft = numSamples - mCurSampleIndex;
-        int32_t framesLeft = (totalSamplesNeeded - samplesProcessed) / numChannels;
-        int32_t numWriteFrames = std::min(framesLeft, samplesLeft / sampleChannels);
-
-        if (numWriteFrames > 0) {
-            const float* data = mSampleBuffer->getSampleData();
-            if ((sampleChannels == 1) && (numChannels == 1)) {
-                // MONO output from MONO samples
-                for (int32_t frameIndex = 0; frameIndex < numWriteFrames; frameIndex++) {
-                    outBuff[samplesProcessed + frameIndex] += data[mCurSampleIndex++] * mGain;
-                }
-            } else if ((sampleChannels == 1) && (numChannels == 2)) {
-                // STEREO output from MONO samples
-                int dstSampleIndex = samplesProcessed;
-                for (int32_t frameIndex = 0; frameIndex < numWriteFrames; frameIndex++) {
-                    outBuff[dstSampleIndex++] += data[mCurSampleIndex] * mLeftGain;
-                    outBuff[dstSampleIndex++] += data[mCurSampleIndex++] * mRightGain;
-                }
-            } else if ((sampleChannels == 2) && (numChannels == 1)) {
-                // MONO output from STEREO samples
-                int dstSampleIndex = samplesProcessed;
-                for (int32_t frameIndex = 0; frameIndex < numWriteFrames; frameIndex++) {
-                    outBuff[dstSampleIndex++] += data[mCurSampleIndex++] * mLeftGain +
-                                                 data[mCurSampleIndex++] * mRightGain;
-                }
-            } else if ((sampleChannels == 2) && (numChannels == 2)) {
-                // STEREO output from STEREO samples
-                int dstSampleIndex = samplesProcessed;
-                for (int32_t frameIndex = 0; frameIndex < numWriteFrames; frameIndex++) {
-                    outBuff[dstSampleIndex++] += data[mCurSampleIndex++] * mLeftGain;
-                    outBuff[dstSampleIndex++] += data[mCurSampleIndex++] * mRightGain;
-                }
-            }
-
-            samplesProcessed += numWriteFrames * numChannels;
-
-            if (mCurSampleIndex >= numSamples) {
-                if (isLoopMode) {
-                    mCurSampleIndex = 0;
-                } else {
-                    mIsPlaying = false;
-                }
-            }
-        } else {
-            break; // No more samples to write in the current chunk
-        }
-    }
-}
-
-} // namespace wavlib