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Delete old tests.

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This commit is contained in:
David Reid
2020-02-22 17:03:04 +10:00
parent 673dce1978
commit 4e91c63874
7 changed files with 0 additions and 2100 deletions
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tests/ma_debug_playback.c
-175
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@@ -1,175 +0,0 @@
#define MA_LOG_LEVEL MA_LOG_LEVEL_VERBOSE
#define MA_DEBUG_OUTPUT
#define MINIAUDIO_IMPLEMENTATION
#include "../miniaudio.h"
int print_context_info(ma_context* pContext)
{
ma_result result = MA_SUCCESS;
ma_device_info* pPlaybackDeviceInfos;
ma_uint32 playbackDeviceCount;
ma_device_info* pCaptureDeviceInfos;
ma_uint32 captureDeviceCount;
printf("BACKEND: %s\n", ma_get_backend_name(pContext->backend));
// Enumeration.
printf(" Enumerating Devices... ");
{
result = ma_context_get_devices(pContext, &pPlaybackDeviceInfos, &playbackDeviceCount, &pCaptureDeviceInfos, &captureDeviceCount);
if (result == MA_SUCCESS) {
printf("Done\n");
} else {
printf("Failed\n");
goto done;
}
printf(" Playback Devices (%d)\n", playbackDeviceCount);
for (ma_uint32 iDevice = 0; iDevice < playbackDeviceCount; ++iDevice) {
printf(" %d: %s\n", iDevice, pPlaybackDeviceInfos[iDevice].name);
}
printf(" Capture Devices (%d)\n", captureDeviceCount);
for (ma_uint32 iDevice = 0; iDevice < captureDeviceCount; ++iDevice) {
printf(" %d: %s\n", iDevice, pCaptureDeviceInfos[iDevice].name);
}
}
// Device Information.
printf(" Getting Device Information...\n");
{
printf(" Playback Devices (%d)\n", playbackDeviceCount);
for (ma_uint32 iDevice = 0; iDevice < playbackDeviceCount; ++iDevice) {
printf(" %d: %s\n", iDevice, pPlaybackDeviceInfos[iDevice].name);
result = ma_context_get_device_info(pContext, ma_device_type_playback, &pPlaybackDeviceInfos[iDevice].id, ma_share_mode_shared, &pPlaybackDeviceInfos[iDevice]);
if (result == MA_SUCCESS) {
printf(" Name: %s\n", pPlaybackDeviceInfos[iDevice].name);
printf(" Min Channels: %d\n", pPlaybackDeviceInfos[iDevice].minChannels);
printf(" Max Channels: %d\n", pPlaybackDeviceInfos[iDevice].maxChannels);
printf(" Min Sample Rate: %d\n", pPlaybackDeviceInfos[iDevice].minSampleRate);
printf(" Max Sample Rate: %d\n", pPlaybackDeviceInfos[iDevice].maxSampleRate);
printf(" Format Count: %d\n", pPlaybackDeviceInfos[iDevice].formatCount);
for (ma_uint32 iFormat = 0; iFormat < pPlaybackDeviceInfos[iDevice].formatCount; ++iFormat) {
printf(" %s\n", ma_get_format_name(pPlaybackDeviceInfos[iDevice].formats[iFormat]));
}
} else {
printf(" ERROR\n");
}
}
printf(" Capture Devices (%d)\n", captureDeviceCount);
for (ma_uint32 iDevice = 0; iDevice < captureDeviceCount; ++iDevice) {
printf(" %d: %s\n", iDevice, pCaptureDeviceInfos[iDevice].name);
result = ma_context_get_device_info(pContext, ma_device_type_capture, &pCaptureDeviceInfos[iDevice].id, ma_share_mode_shared, &pCaptureDeviceInfos[iDevice]);
if (result == MA_SUCCESS) {
printf(" Name: %s\n", pCaptureDeviceInfos[iDevice].name);
printf(" Min Channels: %d\n", pCaptureDeviceInfos[iDevice].minChannels);
printf(" Max Channels: %d\n", pCaptureDeviceInfos[iDevice].maxChannels);
printf(" Min Sample Rate: %d\n", pCaptureDeviceInfos[iDevice].minSampleRate);
printf(" Max Sample Rate: %d\n", pCaptureDeviceInfos[iDevice].maxSampleRate);
printf(" Format Count: %d\n", pCaptureDeviceInfos[iDevice].formatCount);
for (ma_uint32 iFormat = 0; iFormat < pCaptureDeviceInfos[iDevice].formatCount; ++iFormat) {
printf(" %s\n", ma_get_format_name(pCaptureDeviceInfos[iDevice].formats[iFormat]));
}
} else {
printf(" ERROR\n");
}
}
}
done:
printf("\n");
return (result == MA_SUCCESS) ? 0 : -1;
}
int print_device_info(ma_device* pDevice)
{
printf("DEVICE NAME: %s\n", pDevice->name);
printf(" Format: %s -> %s\n", ma_get_format_name(pDevice->format), ma_get_format_name(pDevice->internalFormat));
printf(" Channels: %d -> %d\n", pDevice->channels, pDevice->internalChannels);
printf(" Sample Rate: %d -> %d\n", pDevice->sampleRate, pDevice->internalSampleRate);
printf(" Buffer Size: %d\n", pDevice->bufferSizeInFrames);
printf(" Periods: %d\n", pDevice->periods);
return 0;
}
ma_uint32 on_send(ma_device* pDevice, ma_uint32 frameCount, void* pFramesOut)
{
ma_sine_wave* pSineWave = (ma_sine_wave*)pDevice->pUserData;
ma_assert(pSineWave != NULL);
float* pFramesOutF32 = (float*)pFramesOut;
for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame) {
float sample;
ma_sine_wave_read(pSineWave, 1, &sample);
for (ma_uint32 iChannel = 0; iChannel < pDevice->channels; ++iChannel) {
pFramesOutF32[iChannel] = sample;
}
pFramesOutF32 += pDevice->channels;
}
return frameCount;
}
int main(int argc, char** argv)
{
ma_result result;
ma_sine_wave sineWave;
result = ma_sine_wave_init(0.2, 400, 44100, &sineWave);
if (result != MA_SUCCESS) {
printf("Failed to initialize sine wave.\n");
return -1;
}
// Separate context for this test.
ma_context_config contextConfig = ma_context_config_init(NULL); // <-- Don't need a log callback because we're using debug output instead.
ma_context context;
result = ma_context_init(NULL, 0, &contextConfig, &context);
if (result != MA_SUCCESS) {
printf("Failed to initialize context.\n");
return -1;
}
print_context_info(&context);
// Device.
ma_device_config deviceConfig = ma_device_config_init_playback(ma_format_f32, 2, 44100, on_send);
deviceConfig.bufferSizeInFrames = 32768;
ma_device device;
result = ma_device_init(&context, ma_device_type_playback, NULL, &deviceConfig, &sineWave, &device);
if (result != MA_SUCCESS) {
ma_context_uninit(&context);
printf("Failed to initialize device.\n");
return -1;
}
print_device_info(&device);
// Start playback.
result = ma_device_start(&device);
if (result != MA_SUCCESS) {
ma_device_uninit(&device);
ma_context_uninit(&context);
printf("Failed to start device.\n");
return -1;
}
printf("Press Enter to quit...\n");
getchar();
ma_device_uninit(&device);
ma_context_uninit(&context);
return 0;
}
tests/ma_dithering.c
-148
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@@ -1,148 +0,0 @@
#define MA_DEBUG_OUTPUT
#define MA_USE_REFERENCE_CONVERSION_APIS
#define MINIAUDIO_IMPLEMENTATION
#include "../miniaudio.h"
// Two converters are needed here. One for converting f32 samples from the sine wave generator to the input format,
// and another for converting the input format to the output format for device output.
ma_sine_wave sineWave;
ma_format_converter converterIn;
ma_format_converter converterOut;
ma_uint32 on_convert_samples_in(ma_format_converter* pConverter, ma_uint32 frameCount, void* pFrames, void* pUserData)
{
(void)pUserData;
ma_assert(pConverter->config.formatIn == ma_format_f32);
ma_sine_wave* pSineWave = (ma_sine_wave*)pConverter->config.pUserData;
ma_assert(pSineWave);
return (ma_uint32)ma_sine_wave_read_f32(pSineWave, frameCount, (float*)pFrames);
}
ma_uint32 on_convert_samples_out(ma_format_converter* pConverter, ma_uint32 frameCount, void* pFrames, void* pUserData)
{
(void)pUserData;
ma_format_converter* pConverterIn = (ma_format_converter*)pConverter->config.pUserData;
ma_assert(pConverterIn != NULL);
return (ma_uint32)ma_format_converter_read(pConverterIn, frameCount, pFrames, NULL);
}
void on_send_to_device__original(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
{
ma_assert(pDevice->playback.format == ma_format_f32);
ma_assert(pDevice->playback.channels == 1);
ma_sine_wave_read_f32(&sineWave, frameCount, (float*)pOutput);
(void)pDevice;
(void)pInput;
}
void on_send_to_device__dithered(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
{
ma_assert(pDevice->playback.channels == 1);
ma_format_converter* pConverter = (ma_format_converter*)pDevice->pUserData;
ma_assert(pConverter != NULL);
ma_assert(pDevice->playback.format == pConverter->config.formatOut);
ma_format_converter_read(pConverter, frameCount, pOutput, NULL);
(void)pInput;
}
int do_dithering_test()
{
ma_device_config config;
ma_device device;
ma_result result;
config = ma_device_config_init(ma_device_type_playback);
config.playback.format = ma_format_f32;
config.playback.channels = 1;
config.sampleRate = 0;
config.dataCallback = on_send_to_device__original;
// We first play the sound the way it's meant to be played.
result = ma_device_init(NULL, &config, &device);
if (result != MA_SUCCESS) {
return -1;
}
ma_sine_wave_init(0.5, 400, device.sampleRate, &sineWave);
result = ma_device_start(&device);
if (result != MA_SUCCESS) {
return -2;
}
printf("Press Enter to play enable dithering.\n");
getchar();
ma_device_uninit(&device);
ma_format srcFormat = ma_format_s24;
ma_format dstFormat = ma_format_u8;
ma_dither_mode ditherMode = ma_dither_mode_triangle;
ma_format_converter_config converterInConfig = ma_format_converter_config_init_new();
converterInConfig.formatIn = ma_format_f32; // <-- From the sine wave generator.
converterInConfig.formatOut = srcFormat;
converterInConfig.channels = config.playback.channels;
converterInConfig.ditherMode = ma_dither_mode_none;
converterInConfig.onRead = on_convert_samples_in;
converterInConfig.pUserData = &sineWave;
result = ma_format_converter_init(&converterInConfig, &converterIn);
if (result != MA_SUCCESS) {
return -3;
}
ma_format_converter_config converterOutConfig = ma_format_converter_config_init_new();
converterOutConfig.formatIn = srcFormat;
converterOutConfig.formatOut = dstFormat;
converterOutConfig.channels = config.playback.channels;
converterOutConfig.ditherMode = ditherMode;
converterOutConfig.onRead = on_convert_samples_out;
converterOutConfig.pUserData = &converterIn;
result = ma_format_converter_init(&converterOutConfig, &converterOut);
if (result != MA_SUCCESS) {
return -3;
}
config.playback.format = dstFormat;
config.dataCallback = on_send_to_device__dithered;
config.pUserData = &converterOut;
result = ma_device_init(NULL, &config, &device);
if (result != MA_SUCCESS) {
return -1;
}
// Now we play the sound after it's run through a dithered format converter.
ma_sine_wave_init(0.5, 400, device.sampleRate, &sineWave);
result = ma_device_start(&device);
if (result != MA_SUCCESS) {
return -2;
}
printf("Press Enter to stop.\n");
getchar();
return 0;
}
int main(int argc, char** argv)
{
(void)argc;
(void)argv;
do_dithering_test();
return 0;
}
tests/ma_no_device_io.c
-34
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@@ -1,34 +0,0 @@
// Just a simple test to check that MA_NO_DEVICE_IO compiles.
#include "../extras/dr_flac.h"
#include "../extras/dr_mp3.h"
#include "../extras/dr_wav.h"
#define MA_NO_DEVICE_IO
#define MINIAUDIO_IMPLEMENTATION
#include "../miniaudio.h"
int main(int argc, char** argv)
{
(void)argc;
(void)argv;
ma_result result = MA_ERROR;
ma_pcm_converter_config dspConfig = ma_pcm_converter_config_init_new();
ma_pcm_converter converter;
result = ma_pcm_converter_init(&dspConfig, &converter);
ma_decoder_config decoderConfig = ma_decoder_config_init(ma_format_unknown, 0, 0);
ma_decoder decoder;
result = ma_decoder_init_file("res/sine_s16_mono_48000.wav", &decoderConfig, &decoder);
return result;
}
#define DR_FLAC_IMPLEMENTATION
#include "../extras/dr_flac.h"
#define DR_MP3_IMPLEMENTATION
#include "../extras/dr_mp3.h"
#define DR_WAV_IMPLEMENTATION
#include "../extras/dr_wav.h"
tests/ma_profiling.c
-1172
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tests/ma_resampling.c
-143
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#define MA_NO_SSE2
#define MA_NO_AVX2
#define MINIAUDIO_IMPLEMENTATION
#include "../miniaudio.h"
// There is a usage pattern for resampling that miniaudio does not properly support which is where the client continuously
// reads samples until ma_src_read() returns 0. The problem with this pattern is that is consumes the samples sitting
// in the window which are needed to compute the next samples in future calls to ma_src_read() (assuming the client
// has re-filled the resampler's input data).
/*
for (;;) {
fill_src_input_data(&src, someData);
float buffer[4096]
while ((framesRead = ma_src_read(&src, ...) != 0) {
do_something_with_resampled_data(buffer);
}
}
*/
// In the use case above, the very last samples that are read from ma_src_read() will not have future samples to draw
// from in order to calculate the correct interpolation factor which in turn results in crackling.
ma_uint32 sampleRateIn = 0;
ma_uint32 sampleRateOut = 0;
ma_sine_wave sineWave; // <-- This is the source data.
ma_src src;
float srcInput[1024];
ma_uint32 srcNextSampleIndex = ma_countof(srcInput);
void reload_src_input()
{
ma_sine_wave_read_f32(&sineWave, ma_countof(srcInput), srcInput);
srcNextSampleIndex = 0;
}
ma_uint32 on_src(ma_src* pSRC, ma_uint32 frameCount, void** ppSamplesOut, void* pUserData)
{
ma_assert(pSRC != NULL);
ma_assert(pSRC->config.channels == 1);
(void)pUserData;
// Only read as much as is available in the input buffer. Do not reload the buffer here.
ma_uint32 framesAvailable = ma_countof(srcInput) - srcNextSampleIndex;
ma_uint32 framesToRead = frameCount;
if (framesToRead > framesAvailable) {
framesToRead = framesAvailable;
}
ma_copy_memory(ppSamplesOut[0], srcInput + srcNextSampleIndex, sizeof(float)*framesToRead);
srcNextSampleIndex += framesToRead;
return framesToRead;
}
void on_send_to_device(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
{
(void)pDevice;
(void)pInput;
ma_assert(pDevice->playback.format == ma_format_f32);
ma_assert(pDevice->playback.channels == 1);
float* pFramesF32 = (float*)pOutput;
// To reproduce the case we are needing to test, we need to read from the SRC in a very specific way. We keep looping
// until we've read the requested frame count, however we have an inner loop that keeps running until ma_src_read()
// returns 0, in which case we need to reload the SRC's input data and keep going.
ma_uint32 totalFramesRead = 0;
while (totalFramesRead < frameCount) {
ma_uint32 framesRemaining = frameCount - totalFramesRead;
ma_uint32 maxFramesToRead = 128;
ma_uint32 framesToRead = framesRemaining;
if (framesToRead > maxFramesToRead) {
framesToRead = maxFramesToRead;
}
ma_uint32 framesRead = (ma_uint32)ma_src_read_deinterleaved(&src, framesToRead, (void**)&pFramesF32, NULL);
if (framesRead == 0) {
reload_src_input();
}
totalFramesRead += framesRead;
pFramesF32 += framesRead;
}
ma_assert(totalFramesRead == frameCount);
}
int main(int argc, char** argv)
{
(void)argc;
(void)argv;
ma_device_config config = ma_device_config_init(ma_device_type_playback);
config.playback.format = ma_format_f32;
config.playback.channels = 1;
config.dataCallback = on_send_to_device;
ma_device device;
ma_result result;
config.bufferSizeInFrames = 8192*1;
// We first play the sound the way it's meant to be played.
result = ma_device_init(NULL, &config, &device);
if (result != MA_SUCCESS) {
return -1;
}
// For this test, we need the sine wave to be a different format to the device.
sampleRateOut = device.sampleRate;
sampleRateIn = (sampleRateOut == 44100) ? 48000 : 44100;
ma_sine_wave_init(0.2, 400, sampleRateIn, &sineWave);
ma_src_config srcConfig = ma_src_config_init(sampleRateIn, sampleRateOut, 1, on_src, NULL);
srcConfig.algorithm = ma_src_algorithm_sinc;
srcConfig.neverConsumeEndOfInput = MA_TRUE;
result = ma_src_init(&srcConfig, &src);
if (result != MA_SUCCESS) {
printf("Failed to create SRC.\n");
return -1;
}
result = ma_device_start(&device);
if (result != MA_SUCCESS) {
return -2;
}
printf("Press Enter to quit...\n");
getchar();
ma_device_uninit(&device);
return 0;
}
tests/ma_stop.c
-98
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@@ -1,98 +0,0 @@
#include <stdio.h>
#define MINIAUDIO_IMPLEMENTATION
#include "../miniaudio.h"
ma_sine_wave sineWave;
ma_uint32 framesWritten;
ma_event stopEvent;
ma_bool32 isInitialRun = MA_TRUE;
void on_stop(ma_device* pDevice)
{
(void)pDevice;
printf("STOPPED\n");
}
void on_data(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
{
(void)pInput; /* Not used yet. */
/* Output exactly one second of data. Pad the end with silence. */
ma_uint32 framesRemaining = pDevice->sampleRate - framesWritten;
ma_uint32 framesToProcess = frameCount;
if (framesToProcess > framesRemaining && isInitialRun) {
framesToProcess = framesRemaining;
}
ma_sine_wave_read_f32_ex(&sineWave, framesToProcess, pDevice->playback.channels, ma_stream_layout_interleaved, (float**)&pOutput);
if (isInitialRun) {
framesWritten += framesToProcess;
}
ma_assert(framesWritten <= pDevice->sampleRate);
if (framesWritten >= pDevice->sampleRate) {
if (isInitialRun) {
printf("STOPPING [AUDIO THREAD]...\n");
ma_event_signal(&stopEvent);
isInitialRun = MA_FALSE;
}
}
}
int main(int argc, char** argv)
{
ma_result result;
(void)argc;
(void)argv;
ma_backend backend = ma_backend_wasapi;
ma_sine_wave_init(0.25, 400, 44100, &sineWave);
ma_device_config config = ma_device_config_init(ma_device_type_playback);
config.playback.format = ma_format_f32;
config.playback.channels = 2;
config.sampleRate = 44100;
config.dataCallback = on_data;
config.stopCallback = on_stop;
config.bufferSizeInFrames = 16384;
ma_device device;
result = ma_device_init_ex(&backend, 1, NULL, &config, &device);
if (result != MA_SUCCESS) {
printf("Failed to initialize device.\n");
return result;
}
result = ma_event_init(device.pContext, &stopEvent);
if (result != MA_SUCCESS) {
printf("Failed to initialize stop event.\n");
return result;
}
ma_device_start(&device);
/* We wait for the stop event, stop the device, then ask the user to press any key to restart. This checks that the device can restart after stopping. */
ma_event_wait(&stopEvent);
printf("STOPPING [MAIN THREAD]...\n");
ma_device_stop(&device);
printf("Press Enter to restart...\n");
getchar();
result = ma_device_start(&device);
if (result != MA_SUCCESS) {
printf("Failed to restart the device.\n");
ma_device_uninit(&device);
return -1;
}
printf("Press Enter to quit...\n");
getchar();
ma_device_uninit(&device);
return 0;
}
tests/ma_webaudio_test_0.html
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@@ -1,330 +0,0 @@
<html>
<head>
</head>
<body>
This is a test of the Web Audio API
<div id="Error" style="color:red">
</div>
<div><b>Playback Devices</b></div>
<div id="PlaybackDevices">
[Playback Devices]
</div>
<div><b>Capture Devices</b></div>
<div id="CaptureDevices">
[Capture Devices]
</div>
<button id="btnStartPlayback">
Start Playback
</button>
<button id="btnStopPlayback">
Stop Playback
</button>
<button id="btnClosePlayback">
Close Playback
</button>
<br/>
<button id="btnStartCapture">
Start Capture
</button>
<button id="btnStopCapture">
Stop Capture
</button>
<button id="btnCloseCapture">
Close Capture
</button>
<script>
var runningTime = 0.0;
function ma_enum_devices(deviceType) {
if (deviceType !== 'audiooutput' && deviceType !== 'audioinput') {
alert("Invalid device type: " + deviceType);
return null;
}
// Unfortunately the navigator.mediaDevices.enumerateDevices() API doesn't seem to be very well supported.
// 1) On Chrome and Opera the label is always an empty string
// 2) No devices are returned in Firefox.
//
// This is not a production quality solution right now. It's better to instead just assume default
// devices and output/input silence if they fail to open (if a microphone is not connected, for example).
var promise = new Promise(function(resolve, reject) {
var devices = [];
navigator.mediaDevices.enumerateDevices().then(function(deviceInfos) {
for (var i = 0; i < deviceInfos.length; ++i) {
if (deviceInfos[i].kind === deviceType) {
devices.push(deviceInfos[i]);
}
}
resolve(devices);
}).catch(function(error) {
reject("Failed to enumerate devices: " + error);
});
});
return promise;
}
function ma_device_new(deviceType, deviceID) {
if (typeof(mal) === 'undefined') {
return null; // Context not initialized.
}
// For now only default devices are being used. The device ID needs to be null.
if (deviceID != null) {
return null; // Only default devices are currently supported.
}
if (deviceID == null) {
deviceID = "";
}
var bufferSizeInFrames = 512;
var sampleRate = 44100;
var channelCount = 2;
var device = {};
device.webaudioContext = new (window.AudioContext || window.webkitAudioContext)({
latencyHint: 'interactive',
sampleRate: sampleRate,
});
device.webaudioContext.suspend(); // miniaudio always starts it's devices in a stopped state.
console.log("Sample Rate: " + device.webaudioContext.sampleRate);
device.intermediaryBufferSizeInBytes = channelCount * bufferSizeInFrames * 4;
//device.intermediaryBuffer = Module._malloc(device.intermediaryBufferSizeInBytes);
device.intermediaryBuffer = new Float32Array(channelCount * bufferSizeInFrames);
if (deviceType == 'audiooutput') {
device.playback = {};
device.playback.scriptNode = device.webaudioContext.createScriptProcessor(
bufferSizeInFrames,
channelCount,
channelCount
);
device.playback.scriptNode.onaudioprocess = function(e) {
// TODO: Don't do anything if we don't have an intermediary buffer. This means the device
// was uninitialized.
// The buffer we give to the client needs to be interleaved. After the client callback has returned
// we deinterleave it.
var requiredBufferLength = channelCount * e.outputBuffer.length;
if (device.intermediaryBuffer.length < requiredBufferLength) {
device.intermediaryBuffer = new Float32Array(requiredBufferLength);
}
// Here is where we get the client to fill the buffer with audio data.
// TESTING: Output a sine wave to the speakers.
for (var iFrame = 0; iFrame < e.outputBuffer.length; ++iFrame) {
var value = Math.sin((runningTime+(iFrame*6.28318530717958647693/44100.0)) * 400.0) * 0.25;
for (var iChannel = 0; iChannel < channelCount; ++iChannel) {
device.intermediaryBuffer[iFrame*channelCount + iChannel] = value;
}
}
runningTime += (6.28318530717958647693*e.outputBuffer.length) / 44100.0;
// At this point the intermediary buffer should be filled with data. We now need to deinterleave
// it and write it to the output buffer.
for (var iChannel = 0; iChannel < channelCount; ++iChannel) {
for (var iFrame = 0; iFrame < e.outputBuffer.length; ++iFrame) {
e.outputBuffer.getChannelData(iChannel)[iFrame] = device.intermediaryBuffer[iFrame*channelCount + iChannel];
}
}
};
device.playback.scriptNode.connect(device.webaudioContext.destination);
} else if (deviceType == 'audioinput') {
device.capture = {};
navigator.mediaDevices.getUserMedia({audio:true, video:false})
.then(function(stream) {
// We need to use ScriptProcessorNode instead of MediaRecorder because we need raw PCM data
// rather than compressed data. Why is this not supported? Seriously...
//
// This way this works is that we connect the output of a MediaStreamSourceNode to the input
// of a ScriptProcessorNode. The ScriptProcessorNode is connected to the AudioContext
// destination, but instead of connecting the input to the output we just output silence.
device.capture.streamNode = device.webaudioContext.createMediaStreamSource(stream);
device.capture.scriptNode = device.webaudioContext.createScriptProcessor(
bufferSizeInFrames,
channelCount,
channelCount
);
device.capture.scriptNode.onaudioprocess = function(e) {
// The input buffer needs to be interleaved before sending to the client. We need to do
// this in an intermediary buffer.
var requiredBufferLength = e.inputBuffer.numberOfChannels * e.inputBuffer.length;
if (device.intermediaryBuffer.length < requiredBufferLength) {
device.intermediaryBuffer = new Float32Array(requiredBufferLength);
}
for (var iFrame = 0; iFrame < e.inputBuffer.length; ++iFrame) {
for (var iChannel = 0; iChannel < e.inputBuffer.numberOfChannels; ++iChannel) {
device.intermediaryBuffer[iFrame*e.inputBuffer.numberOfChannels + iChannel] = e.inputBuffer.getChannelData(iChannel)[iFrame];
}
}
// At this point the input data has been interleaved and can be passed on to the client.
// Always output silence.
for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
e.outputBuffer.getChannelData(iChannel).fill(0.0);
}
/*
// TESTING: Write to the interleaved data to the output buffers.
for (var iChannel = 0; iChannel < e.inputBuffer.numberOfChannels; ++iChannel) {
for (var iFrame = 0; iFrame < e.inputBuffer.length; ++iFrame) {
e.outputBuffer.getChannelData(iChannel)[iFrame] = device.intermediaryBuffer[iFrame*e.inputBuffer.numberOfChannels + iChannel];
}
}
*/
};
device.capture.streamNode.connect(device.capture.scriptNode);
device.capture.scriptNode.connect(device.webaudioContext.destination);
})
.catch(function(error) {
// For now just do nothing, but later on we may want to periodically fire the callback with silence.
console.log("No Stream.");
});
} else {
return null; // Unknown device type.
}
return device;
}
function ma_device_delete(device) {
Module._free(device.intermediaryBuffer);
}
function ma_context_init() {
if ((window.AudioContext || window.webkitAudioContext) === undefined) {
return 0; // Web Audio not supported.
}
if (typeof(Float32Array) === 'undefined') {
return 0; // Float32Array not supported.
}
if (typeof(mal) === 'undefined') {
mal = {};
miniaudio.devices = []; // Device cache for mapping devices to indexes for JavaScript/C interop.
// Returns the index of the device. Throws an exception on error.
miniaudio.track_device = function(device) {
// Try inserting into a free slot first.
for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) {
if (miniaudio.devices[iDevice] == null) {
miniaudio.devices[iDevice] = device;
return iDevice;
}
}
// Getting here means there is no empty slots in the array so we just push to the end.
miniaudio.devices.push(device);
return miniaudio.devices.length - 1;
};
miniaudio.untrack_device_by_index = function(deviceIndex) {
// We just set the device's slot to null. The slot will get reused in the next call to ma_track_device.
miniaudio.devices[iDevice] = null;
// Trim the array if possible.
while (miniaudio.devices.length > 0) {
if (miniaudio.devices[miniaudio.devices.length-1] == null) {
miniaudio.devices.pop();
} else {
break;
}
}
};
miniaudio.untrack_device = function(device) {
for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) {
if (miniaudio.devices[iDevice] == device) {
return miniaudio.untrack_device_by_index(iDevice);
}
}
};
miniaudio.get_device_by_index = function(deviceIndex) {
return miniaudio.devices[deviceIndex];
};
}
return 1;
}
window.onload = function() {
if (ma_context_init() != 1) {
alert("Failed to initialize context.");
return;
}
// Unfortunately this doesn't seem to work too well. See comment in ma_enum_devices().
ma_enum_devices('audiooutput').then(function(outputDevices) {
for (var iDevice = 0; iDevice < outputDevices.length; ++iDevice) {
console.log("Output Device: ", JSON.stringify(outputDevices[iDevice]));
}
}).catch(function(error) {
console.log("Failed to retrieve output devices: ", error);
});
ma_enum_devices('audioinput').then(function(inputDevices) {
for (var iDevice = 0; iDevice < inputDevices.length; ++iDevice) {
console.log("Input Device: ", JSON.stringify(inputDevices[iDevice]));
}
}).catch(function(error) {
console.log("Failed to retrieve input devices: ", error);
});
var outputDevice = ma_device_new('audiooutput', null);
var inputDevice = ma_device_new('audioinput', null);
var btnStartPlayback = document.getElementById("btnStartPlayback");
btnStartPlayback.addEventListener('click', function() {
outputDevice.webaudioContext.resume();
});
var btnStopPlayback = document.getElementById("btnStopPlayback");
btnStopPlayback.addEventListener('click', function() {
outputDevice.webaudioContext.suspend();
});
var btnClosePlayback = document.getElementById("btnClosePlayback");
btnClosePlayback.addEventListener('click', function() {
outputDevice.webaudioContext.close();
});
var btnStartCapture = document.getElementById("btnStartCapture");
btnStartCapture.addEventListener('click', function() {
inputDevice.webaudioContext.resume();
});
var btnStopCapture = document.getElementById("btnStopCapture");
btnStopCapture.addEventListener('click', function() {
inputDevice.webaudioContext.suspend();
});
var btnCloseCapture = document.getElementById("btnCloseCapture");
btnCloseCapture.addEventListener('click', function() {
inputDevice.webaudioContext.close();
});
}
</script>
</body>
</html>