int RtInOut( void* outputBuffer, void* inputBuffer, unsigned int framesPerBuffer,
double streamTime, RtAudioStreamStatus status, void *userdata )
{
if(!UNTZ::System::get()->getData()->isActive())
{
memset(outputBuffer, 0, sizeof(float) * framesPerBuffer * UNTZ::System::get()->getData()->getNumOutputChannels());
return 0;
}
if(status)
std::cout << "Stream underflow detected!" << std::endl;
AudioMixer *mixer = (AudioMixer*)userdata;
mixer->process(0, NULL, UNTZ::System::get()->getData()->getNumOutputChannels(), (float*)outputBuffer, framesPerBuffer);
// volume & clipping
// HBS
UInt32 samples = UNTZ::System::get()->getData()->getNumOutputChannels() * framesPerBuffer;
float volume = mixer->getVolume();
// TODO: doing an extra read/write here is painful...
float *outB = (float*)outputBuffer;
for (UInt32 k = 0; k < samples; ++k)
{
float val = *outB * volume;
val = val > 1.0 ? 1.0 : val;
val = val < -1.0 ? -1.0 : val;
*(outB)++ = val;
}
return 0;
}
int RtInOut( void* outputBuffer, void* inputBuffer, unsigned int framesPerBuffer,
double streamTime, RtAudioStreamStatus status, void *userdata )
{
LinuxSystemData *sysData = (LinuxSystemData *)userdata;
UInt32 numOutputChannels = UNTZ::System::get()->getData()->getNumOutputChannels();
UInt32 samples = numOutputChannels * framesPerBuffer;
if(!UNTZ::System::get()->getData()->isActive())
{
memset(outputBuffer, 0, sizeof(float) * samples);
return 0;
}
if(sysData->mOutputBuffer.size() < samples)
{
sysData->mOutputBuffer.resize(samples);
}
float *mixerOutputBuffer = (float*)&sysData->mOutputBuffer[0];
if(status)
std::cout << "Stream underflow detected!" << std::endl;
AudioMixer *mixer = &sysData->mMixer;
mixer->process(0, NULL, numOutputChannels, mixerOutputBuffer, framesPerBuffer);
// volume & clipping & interleaving
float volume = mixer->getVolume();
float *outB = (float*)outputBuffer;
for(UInt32 i=0; i<framesPerBuffer; i++)
{
for(UInt32 j=0; j<numOutputChannels; j++)
{
float val = volume * mixerOutputBuffer[j*framesPerBuffer+i];
val = val > 1.0 ? 1.0 : val;
val = val < -1.0 ? -1.0 : val;
*(outB)++ = val;
}
}
return 0;
}
int main(int argc, char* argv[]) {
const char* const progname = argv[0];
bool useInputFloat = false;
bool useMixerFloat = false;
bool useRamp = true;
uint32_t outputSampleRate = 48000;
uint32_t outputChannels = 2; // stereo for now
std::vector<int> Pvalues;
const char* outputFilename = NULL;
const char* auxFilename = NULL;
std::vector<int32_t> names;
std::vector<SignalProvider> providers;
std::vector<audio_format_t> formats;
for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) {
switch (ch) {
case 'f':
useInputFloat = true;
break;
case 'm':
useMixerFloat = true;
break;
case 'c':
outputChannels = atoi(optarg);
break;
case 's':
outputSampleRate = atoi(optarg);
break;
case 'o':
outputFilename = optarg;
break;
case 'a':
auxFilename = optarg;
break;
case 'P':
if (parseCSV(optarg, Pvalues) < 0) {
fprintf(stderr, "incorrect syntax for -P option\n");
return EXIT_FAILURE;
}
break;
case '?':
default:
usage(progname);
return EXIT_FAILURE;
}
}
argc -= optind;
argv += optind;
if (argc == 0) {
usage(progname);
return EXIT_FAILURE;
}
size_t outputFrames = 0;
// create providers for each track
names.resize(argc);
providers.resize(argc);
formats.resize(argc);
for (int i = 0; i < argc; ++i) {
static const char chirp[] = "chirp:";
static const char sine[] = "sine:";
static const double kSeconds = 1;
bool useFloat = useInputFloat;
if (!strncmp(argv[i], chirp, strlen(chirp))) {
std::vector<int> v;
const char *s = parseFormat(argv[i] + strlen(chirp), &useFloat);
parseCSV(s, v);
if (v.size() == 2) {
printf("creating chirp(%d %d)\n", v[0], v[1]);
if (useFloat) {
providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
providers[i].setIncr(Pvalues);
} else {
fprintf(stderr, "malformed input '%s'\n", argv[i]);
}
} else if (!strncmp(argv[i], sine, strlen(sine))) {
std::vector<int> v;
const char *s = parseFormat(argv[i] + strlen(sine), &useFloat);
parseCSV(s, v);
if (v.size() == 3) {
printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]);
if (useFloat) {
providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
providers[i].setIncr(Pvalues);
} else {
fprintf(stderr, "malformed input '%s'\n", argv[i]);
}
} else {
printf("creating filename(%s)\n", argv[i]);
if (useInputFloat) {
providers[i].setFile<float>(argv[i]);
formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
providers[i].setFile<short>(argv[i]);
formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
providers[i].setIncr(Pvalues);
}
// calculate the number of output frames
size_t nframes = (int64_t) providers[i].getNumFrames() * outputSampleRate
/ providers[i].getSampleRate();
if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames
outputFrames = nframes;
}
}
// create the output buffer.
const size_t outputFrameSize = outputChannels
* (useMixerFloat ? sizeof(float) : sizeof(int16_t));
const size_t outputSize = outputFrames * outputFrameSize;
const audio_channel_mask_t outputChannelMask =
audio_channel_out_mask_from_count(outputChannels);
void *outputAddr = NULL;
(void) posix_memalign(&outputAddr, 32, outputSize);
memset(outputAddr, 0, outputSize);
// create the aux buffer, if needed.
const size_t auxFrameSize = sizeof(int32_t); // Q4.27 always
const size_t auxSize = outputFrames * auxFrameSize;
void *auxAddr = NULL;
if (auxFilename) {
(void) posix_memalign(&auxAddr, 32, auxSize);
memset(auxAddr, 0, auxSize);
}
// create the mixer.
const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960
AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate);
audio_format_t mixerFormat = useMixerFloat
? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
float f = AudioMixer::UNITY_GAIN_FLOAT / providers.size(); // normalize volume by # tracks
static float f0; // zero
// set up the tracks.
for (size_t i = 0; i < providers.size(); ++i) {
//printf("track %d out of %d\n", i, providers.size());
uint32_t channelMask = audio_channel_out_mask_from_count(providers[i].getNumChannels());
const int name = i;
const status_t status = mixer->create(
name, channelMask, formats[i], AUDIO_SESSION_OUTPUT_MIX);
LOG_ALWAYS_FATAL_IF(status != OK);
names[i] = name;
mixer->setBufferProvider(name, &providers[i]);
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(void *)outputAddr);
mixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MIXER_FORMAT,
(void *)(uintptr_t)mixerFormat);
mixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::FORMAT,
(void *)(uintptr_t)formats[i]);
mixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MIXER_CHANNEL_MASK,
(void *)(uintptr_t)outputChannelMask);
mixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::CHANNEL_MASK,
(void *)(uintptr_t)channelMask);
mixer->setParameter(
name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(uintptr_t)providers[i].getSampleRate());
if (useRamp) {
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0);
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0);
mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME0, &f);
mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME1, &f);
} else {
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f);
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f);
}
if (auxFilename) {
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
(void *) auxAddr);
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::AUXLEVEL, &f0);
mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::AUXLEVEL, &f);
}
mixer->enable(name);
}
// pump the mixer to process data.
size_t i;
for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) {
for (size_t j = 0; j < names.size(); ++j) {
mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(char *) outputAddr + i * outputFrameSize);
if (auxFilename) {
mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
(char *) auxAddr + i * auxFrameSize);
}
}
mixer->process();
}
outputFrames = i; // reset output frames to the data actually produced.
// write to files
writeFile(outputFilename, outputAddr,
outputSampleRate, outputChannels, outputFrames, useMixerFloat);
if (auxFilename) {
// Aux buffer is always in q4_27 format for O and earlier.
// memcpy_to_i16_from_q4_27((int16_t*)auxAddr, (const int32_t*)auxAddr, outputFrames);
// Aux buffer is always in float format for P.
memcpy_to_i16_from_float((int16_t*)auxAddr, (const float*)auxAddr, outputFrames);
writeFile(auxFilename, auxAddr, outputSampleRate, 1, outputFrames, false);
}
delete mixer;
free(outputAddr);
free(auxAddr);
return EXIT_SUCCESS;
}
