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Ref T730, namespace for BlackCore::Afv

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* added namespace
* removed some *.pri files and added files to blackcore.pro
* added copyright etc.
This commit is contained in:
Klaus Basan
2019-09-19 00:56:54 +02:00
committed by Mat Sutcliffe
parent d064da13b5
commit 384aa3ce19
46 changed files with 2985 additions and 2916 deletions
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264
src/blackcore/afv/audio/input.cpp
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@@ -1,3 +1,13 @@
/* Copyright (C) 2019
* swift project Community / Contributors
*
* This file is part of swift project. It is subject to the license terms in the LICENSE file found in the top-level
* directory of this distribution. No part of swift project, including this file, may be copied, modified, propagated,
* or distributed except according to the terms contained in the LICENSE file.
*/
//! \file
#include "input.h"
#include "blacksound/audioutilities.h"
@@ -5,148 +15,158 @@
#include <QDebug>
#include <cmath>
void AudioInputBuffer::start()
namespace BlackCore
{
open(QIODevice::WriteOnly);
}
void AudioInputBuffer::stop()
{
close();
}
qint64 AudioInputBuffer::readData(char *data, qint64 maxlen)
{
Q_UNUSED(data)
Q_UNUSED(maxlen)
return 0;
}
qint64 AudioInputBuffer::writeData(const char *data, qint64 len)
{
QByteArray buffer(data, static_cast<int>(len));
m_buffer.append(buffer);
// 20 ms = 960 samples * 2 bytes = 1920 Bytes
if (m_buffer.size() >= 1920)
namespace Afv
{
emit frameAvailable(m_buffer.left(1920));
m_buffer.remove(0, 1920);
}
namespace Audio
{
void AudioInputBuffer::start()
{
open(QIODevice::WriteOnly);
}
return len;
}
void AudioInputBuffer::stop()
{
close();
}
Input::Input(int sampleRate, QObject *parent) :
QObject(parent),
m_sampleRate(sampleRate),
m_encoder(sampleRate, 1, OPUS_APPLICATION_VOIP)
{
m_encoder.setBitRate(16 * 1024);
}
qint64 AudioInputBuffer::readData(char *data, qint64 maxlen)
{
Q_UNUSED(data)
Q_UNUSED(maxlen)
bool Input::started() const
{
return m_started;
}
return 0;
}
int Input::opusBytesEncoded() const
{
return m_opusBytesEncoded;
}
qint64 AudioInputBuffer::writeData(const char *data, qint64 len)
{
QByteArray buffer(data, static_cast<int>(len));
m_buffer.append(buffer);
// 20 ms = 960 samples * 2 bytes = 1920 Bytes
if (m_buffer.size() >= 1920)
{
emit frameAvailable(m_buffer.left(1920));
m_buffer.remove(0, 1920);
}
void Input::setOpusBytesEncoded(int opusBytesEncoded)
{
m_opusBytesEncoded = opusBytesEncoded;
}
return len;
}
float Input::volume() const
{
return m_volume;
}
Input::Input(int sampleRate, QObject *parent) :
QObject(parent),
m_sampleRate(sampleRate),
m_encoder(sampleRate, 1, OPUS_APPLICATION_VOIP)
{
m_encoder.setBitRate(16 * 1024);
}
void Input::setVolume(float volume)
{
m_volume = volume;
}
bool Input::started() const
{
return m_started;
}
void Input::start(const QAudioDeviceInfo &inputDevice)
{
if (m_started) { return; }
int Input::opusBytesEncoded() const
{
return m_opusBytesEncoded;
}
QAudioFormat waveFormat;
void Input::setOpusBytesEncoded(int opusBytesEncoded)
{
m_opusBytesEncoded = opusBytesEncoded;
}
waveFormat.setSampleRate(m_sampleRate);
waveFormat.setChannelCount(1);
waveFormat.setSampleSize(16);
waveFormat.setSampleType(QAudioFormat::SignedInt);
waveFormat.setByteOrder(QAudioFormat::LittleEndian);
waveFormat.setCodec("audio/pcm");
float Input::volume() const
{
return m_volume;
}
QAudioFormat inputFormat = waveFormat;
if (!inputDevice.isFormatSupported(inputFormat))
{
qWarning() << "Default format not supported - trying to use nearest";
inputFormat = inputDevice.nearestFormat(inputFormat);
}
void Input::setVolume(float volume)
{
m_volume = volume;
}
m_audioInput.reset(new QAudioInput(inputDevice, inputFormat));
// We want 20 ms of buffer size
// 20 ms * nSamplesPerSec × nChannels × wBitsPerSample / 8 x 1000
int bufferSize = 20 * inputFormat.sampleRate() * inputFormat.channelCount() * inputFormat.sampleSize() / ( 8 * 1000 );
m_audioInput->setBufferSize(bufferSize);
m_audioInputBuffer.start();
m_audioInput->start(&m_audioInputBuffer);
connect(&m_audioInputBuffer, &AudioInputBuffer::frameAvailable, this, &Input::audioInDataAvailable);
void Input::start(const QAudioDeviceInfo &inputDevice)
{
if (m_started) { return; }
m_started = true;
}
QAudioFormat waveFormat;
void Input::stop()
{
if (! m_started) { return; }
waveFormat.setSampleRate(m_sampleRate);
waveFormat.setChannelCount(1);
waveFormat.setSampleSize(16);
waveFormat.setSampleType(QAudioFormat::SignedInt);
waveFormat.setByteOrder(QAudioFormat::LittleEndian);
waveFormat.setCodec("audio/pcm");
m_started = false;
QAudioFormat inputFormat = waveFormat;
if (!inputDevice.isFormatSupported(inputFormat))
{
qWarning() << "Default format not supported - trying to use nearest";
inputFormat = inputDevice.nearestFormat(inputFormat);
}
m_audioInput->stop();
m_audioInput.reset();
}
m_audioInput.reset(new QAudioInput(inputDevice, inputFormat));
// We want 20 ms of buffer size
// 20 ms * nSamplesPerSec × nChannels × wBitsPerSample / 8 x 1000
int bufferSize = 20 * inputFormat.sampleRate() * inputFormat.channelCount() * inputFormat.sampleSize() / (8 * 1000);
m_audioInput->setBufferSize(bufferSize);
m_audioInputBuffer.start();
m_audioInput->start(&m_audioInputBuffer);
connect(&m_audioInputBuffer, &AudioInputBuffer::frameAvailable, this, &Input::audioInDataAvailable);
void Input::audioInDataAvailable(const QByteArray &frame)
{
const QVector<qint16> samples = convertBytesTo16BitPCM(frame);
m_started = true;
}
int length;
QByteArray encodedBuffer = m_encoder.encode(samples, samples.size(), &length);
m_opusBytesEncoded += length;
void Input::stop()
{
if (! m_started) { return; }
for (const qint16 sample : samples)
{
qint16 sampleInput = sample;
sampleInput = qAbs(sampleInput);
if (sampleInput > m_maxSampleInput)
m_maxSampleInput = sampleInput;
}
m_started = false;
m_sampleCount += samples.size();
if (m_sampleCount >= c_sampleCountPerEvent)
{
InputVolumeStreamArgs inputVolumeStreamArgs;
qint16 maxInt = std::numeric_limits<qint16>::max();
inputVolumeStreamArgs.PeakRaw = m_maxSampleInput / maxInt;
inputVolumeStreamArgs.PeakDB = (float)(20 * std::log10(inputVolumeStreamArgs.PeakRaw));
float db = qBound(minDb, inputVolumeStreamArgs.PeakDB, maxDb);
float ratio = (db - minDb) / (maxDb - minDb);
if (ratio < 0.30)
ratio = 0;
if (ratio > 1.0)
ratio = 1;
inputVolumeStreamArgs.PeakVU = ratio;
emit inputVolumeStream(inputVolumeStreamArgs);
m_sampleCount = 0;
m_maxSampleInput = 0;
}
m_audioInput->stop();
m_audioInput.reset();
}
OpusDataAvailableArgs opusDataAvailableArgs = { m_audioSequenceCounter++, encodedBuffer };
emit opusDataAvailable(opusDataAvailableArgs);
}
void Input::audioInDataAvailable(const QByteArray &frame)
{
const QVector<qint16> samples = convertBytesTo16BitPCM(frame);
int length;
QByteArray encodedBuffer = m_encoder.encode(samples, samples.size(), &length);
m_opusBytesEncoded += length;
for (const qint16 sample : samples)
{
qint16 sampleInput = sample;
sampleInput = qAbs(sampleInput);
if (sampleInput > m_maxSampleInput)
m_maxSampleInput = sampleInput;
}
m_sampleCount += samples.size();
if (m_sampleCount >= c_sampleCountPerEvent)
{
InputVolumeStreamArgs inputVolumeStreamArgs;
qint16 maxInt = std::numeric_limits<qint16>::max();
inputVolumeStreamArgs.PeakRaw = m_maxSampleInput / maxInt;
inputVolumeStreamArgs.PeakDB = (float)(20 * std::log10(inputVolumeStreamArgs.PeakRaw));
float db = qBound(minDb, inputVolumeStreamArgs.PeakDB, maxDb);
float ratio = (db - minDb) / (maxDb - minDb);
if (ratio < 0.30)
ratio = 0;
if (ratio > 1.0)
ratio = 1;
inputVolumeStreamArgs.PeakVU = ratio;
emit inputVolumeStream(inputVolumeStreamArgs);
m_sampleCount = 0;
m_maxSampleInput = 0;
}
OpusDataAvailableArgs opusDataAvailableArgs = { m_audioSequenceCounter++, encodedBuffer };
emit opusDataAvailable(opusDataAvailableArgs);
}
} // ns
} // ns
} // ns