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cleanup the callers of the resamplers for mic to net and net to speaker, remove loopback resampler

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This commit is contained in:
Brad Hefta-Gaub 2016-06-09 13:28:25 -07:00
parent a48cce2975
commit c0be9aec6a
2 changed files with 26 additions and 63 deletions
libraries/audio-client/src
AudioClient.cppAudioClient.h

88
libraries/audio-client/src/AudioClient.cpp
View file

@ -102,7 +102,6 @@ AudioClient::AudioClient() :
_reverbOptions(&_scriptReverbOptions),
_inputToNetworkResampler(NULL),
_networkToOutputResampler(NULL),
_loopbackResampler(NULL),
_outgoingAvatarAudioSequenceNumber(0),
_audioOutputIODevice(_receivedAudioStream, this),
_stats(&_receivedAudioStream),
@ -315,54 +314,35 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
// FIXME: directly using 24khz has a bug somewhere that causes channels to be swapped.
// Continue using our internal resampler, for now.
if (true || !audioDevice.isFormatSupported(desiredAudioFormat)) {
qCDebug(audioclient) << "The desired format for audio I/O is" << desiredAudioFormat;
qCDebug(audioclient, "The desired audio format is not supported by this device");
qCDebug(audioclient) << "The desired format for audio I/O is" << desiredAudioFormat;
if (desiredAudioFormat.channelCount() == 1) {
adjustedAudioFormat = desiredAudioFormat;
adjustedAudioFormat.setChannelCount(2);
if (false && audioDevice.isFormatSupported(adjustedAudioFormat)) {
return true;
} else {
adjustedAudioFormat.setChannelCount(1);
}
}
const int FORTY_FOUR = 44100;
adjustedAudioFormat = desiredAudioFormat;
const int FORTY_FOUR = 44100;
adjustedAudioFormat = desiredAudioFormat;
#ifdef Q_OS_ANDROID
adjustedAudioFormat.setSampleRate(FORTY_FOUR);
adjustedAudioFormat.setSampleRate(FORTY_FOUR);
#else
const int HALF_FORTY_FOUR = FORTY_FOUR / 2;
const int HALF_FORTY_FOUR = FORTY_FOUR / 2;
if (audioDevice.supportedSampleRates().contains(AudioConstants::SAMPLE_RATE * 2)) {
// use 48, which is a sample downsample, upsample
adjustedAudioFormat.setSampleRate(AudioConstants::SAMPLE_RATE * 2);
} else if (audioDevice.supportedSampleRates().contains(HALF_FORTY_FOUR)) {
// use 22050, resample but closer to 24
adjustedAudioFormat.setSampleRate(HALF_FORTY_FOUR);
} else if (audioDevice.supportedSampleRates().contains(FORTY_FOUR)) {
// use 48000, resample
adjustedAudioFormat.setSampleRate(FORTY_FOUR);
}
if (audioDevice.supportedSampleRates().contains(AudioConstants::SAMPLE_RATE * 2)) {
// use 48, which is a sample downsample, upsample
adjustedAudioFormat.setSampleRate(AudioConstants::SAMPLE_RATE * 2);
} else if (audioDevice.supportedSampleRates().contains(HALF_FORTY_FOUR)) {
// use 22050, resample but closer to 24
adjustedAudioFormat.setSampleRate(HALF_FORTY_FOUR);
} else if (audioDevice.supportedSampleRates().contains(FORTY_FOUR)) {
// use 48000, resample
adjustedAudioFormat.setSampleRate(FORTY_FOUR);
}
#endif
if (adjustedAudioFormat != desiredAudioFormat) {
// return the nearest in case it needs 2 channels
adjustedAudioFormat = audioDevice.nearestFormat(adjustedAudioFormat);
return true;
} else {
return false;
}
} else {
// set the adjustedAudioFormat to the desiredAudioFormat, since it will work
adjustedAudioFormat = desiredAudioFormat;
if (adjustedAudioFormat != desiredAudioFormat) {
// return the nearest in case it needs 2 channels
adjustedAudioFormat = audioDevice.nearestFormat(adjustedAudioFormat);
return true;
} else {
return false;
}
}
@ -467,11 +447,6 @@ void AudioClient::stop() {
// "switch" to invalid devices in order to shut down the state
switchInputToAudioDevice(QAudioDeviceInfo());
switchOutputToAudioDevice(QAudioDeviceInfo());
if (_loopbackResampler) {
delete _loopbackResampler;
_loopbackResampler = NULL;
}
}
void AudioClient::handleAudioEnvironmentDataPacket(QSharedPointer<ReceivedMessage> message) {
@ -675,16 +650,10 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
}
}
// do we need to setup a resampler?
if (_inputFormat.sampleRate() != _outputFormat.sampleRate() && !_loopbackResampler) {
qCDebug(audioclient) << "Attemping to create a resampler for input format to output format for audio loopback.";
assert(_inputFormat.sampleSize() == 16);
assert(_outputFormat.sampleSize() == 16);
int channelCount = (_inputFormat.channelCount() == 2 && _outputFormat.channelCount() == 2) ? 2 : 1;
_loopbackResampler = new AudioSRC(_inputFormat.sampleRate(), _outputFormat.sampleRate(), channelCount);
}
// NOTE: we assume the inputFormat and the outputFormat are the same, since on any modern
// multimedia OS they should be. If there is a device that this is not true for, we can
// add back support to do resampling.
Q_ASSERT(_inputFormat.sampleRate() == _outputFormat.sampleRate());
static QByteArray loopBackByteArray;
@ -696,7 +665,8 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
int16_t* inputSamples = reinterpret_cast<int16_t*>(inputByteArray.data());
int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
possibleResampling(_loopbackResampler,
auto NO_RESAMPLER = nullptr;
possibleResampling(NO_RESAMPLER,
inputSamples, loopbackSamples,
numInputSamples, numLoopbackSamples,
_inputFormat, _outputFormat);
@ -1039,12 +1009,6 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
_networkToOutputResampler = NULL;
}
if (_loopbackResampler) {
// if we were using an input to output resample, delete it here
delete _loopbackResampler;
_loopbackResampler = NULL;
}
if (!outputDeviceInfo.isNull()) {
qCDebug(audioclient) << "The audio output device " << outputDeviceInfo.deviceName() << "is available.";
_outputAudioDeviceName = outputDeviceInfo.deviceName().trimmed();

1
libraries/audio-client/src/AudioClient.h
View file

@ -260,7 +260,6 @@ private:
// possible streams needed for resample
AudioSRC* _inputToNetworkResampler;
AudioSRC* _networkToOutputResampler;
AudioSRC* _loopbackResampler;
// Adds Reverb
void configureReverb();