From c0be9aec6aedd683e12d7d61dfbf9ff210526d70 Mon Sep 17 00:00:00 2001 From: Brad Hefta-Gaub Date: Thu, 9 Jun 2016 13:28:25 -0700 Subject: [PATCH] cleanup the callers of the resamplers for mic to net and net to speaker, remove loopback resampler --- libraries/audio-client/src/AudioClient.cpp | 88 +++++++--------------- libraries/audio-client/src/AudioClient.h | 1 - 2 files changed, 26 insertions(+), 63 deletions(-) diff --git a/libraries/audio-client/src/AudioClient.cpp b/libraries/audio-client/src/AudioClient.cpp index 0c7a79e2a3..ac055bb0f6 100644 --- a/libraries/audio-client/src/AudioClient.cpp +++ b/libraries/audio-client/src/AudioClient.cpp @@ -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 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(inputByteArray.data()); int16_t* loopbackSamples = reinterpret_cast(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(); diff --git a/libraries/audio-client/src/AudioClient.h b/libraries/audio-client/src/AudioClient.h index 3a14c878f6..dc46db5657 100644 --- a/libraries/audio-client/src/AudioClient.h +++ b/libraries/audio-client/src/AudioClient.h @@ -260,7 +260,6 @@ private: // possible streams needed for resample AudioSRC* _inputToNetworkResampler; AudioSRC* _networkToOutputResampler; - AudioSRC* _loopbackResampler; // Adds Reverb void configureReverb();