Clean-up; headers added

preferencesAboutTextFull
Craig Watson 9 years ago
parent 1e731be171
commit 89bd259d60
  1. 30
      src/podcast/ffmpeg/UBFFmpegVideoEncoder.cpp
  2. 21
      src/podcast/ffmpeg/UBFFmpegVideoEncoder.h
  3. 177
      src/podcast/ffmpeg/UBMicrophoneInput.cpp
  4. 29
      src/podcast/ffmpeg/UBMicrophoneInput.h

@ -1,3 +1,24 @@
/*
* Copyright (C) 2015-2016 Département de l'Instruction Publique (DIP-SEM)
*
* This file is part of OpenBoard.
*
* OpenBoard is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, version 3 of the License,
* with a specific linking exception for the OpenSSL project's
* "OpenSSL" library (or with modified versions of it that use the
* same license as the "OpenSSL" library).
*
* OpenBoard is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with OpenBoard. If not, see <http://www.gnu.org/licenses/>.
*/
#include "UBFFmpegVideoEncoder.h" #include "UBFFmpegVideoEncoder.h"
//------------------------------------------------------------------------- //-------------------------------------------------------------------------
@ -223,11 +244,6 @@ bool UBFFmpegVideoEncoder::init()
int inChannelCount = mAudioInput->channelCount(); int inChannelCount = mAudioInput->channelCount();
int inSampleRate = mAudioInput->sampleRate(); int inSampleRate = mAudioInput->sampleRate();
int inSampleSize = mAudioInput->sampleSize();
qDebug() << "inChannelCount = " << inChannelCount;
qDebug() << "inSampleRate = " << inSampleRate;
qDebug() << "inSampleSize = " << inSampleSize;
// Codec // Codec
@ -535,8 +551,8 @@ void UBFFmpegVideoEncoderWorker::queueAudioFrame(AVFrame* frame)
} }
/** /**
* The main encoding function. Takes the queued image frames and * The main encoding function. Takes the queued frames and
* assembles them into the video * writes them to the video and audio streams
*/ */
void UBFFmpegVideoEncoderWorker::runEncoding() void UBFFmpegVideoEncoderWorker::runEncoding()
{ {

@ -1,3 +1,24 @@
/*
* Copyright (C) 2015-2016 Département de l'Instruction Publique (DIP-SEM)
*
* This file is part of OpenBoard.
*
* OpenBoard is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, version 3 of the License,
* with a specific linking exception for the OpenSSL project's
* "OpenSSL" library (or with modified versions of it that use the
* same license as the "OpenSSL" library).
*
* OpenBoard is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with OpenBoard. If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef UBFFMPEGVIDEOENCODER_H #ifndef UBFFMPEGVIDEOENCODER_H
#define UBFFMPEGVIDEOENCODER_H #define UBFFMPEGVIDEOENCODER_H

@ -1,3 +1,24 @@
/*
* Copyright (C) 2015-2016 Département de l'Instruction Publique (DIP-SEM)
*
* This file is part of OpenBoard.
*
* OpenBoard is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, version 3 of the License,
* with a specific linking exception for the OpenSSL project's
* "OpenSSL" library (or with modified versions of it that use the
* same license as the "OpenSSL" library).
*
* OpenBoard is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with OpenBoard. If not, see <http://www.gnu.org/licenses/>.
*/
#include "UBMicrophoneInput.h" #include "UBMicrophoneInput.h"
UBMicrophoneInput::UBMicrophoneInput() UBMicrophoneInput::UBMicrophoneInput()
@ -13,80 +34,6 @@ UBMicrophoneInput::~UBMicrophoneInput()
delete mAudioInput; delete mAudioInput;
} }
int UBMicrophoneInput::channelCount()
{
return mAudioFormat.channelCount();
}
int UBMicrophoneInput::sampleRate()
{
return mAudioFormat.sampleRate();
}
/* Return the sample size in bits */
int UBMicrophoneInput::sampleSize()
{
return mAudioFormat.sampleSize();
}
/** Return the sample format in FFMpeg style (AVSampleFormat enum) */
int UBMicrophoneInput::sampleFormat()
{
enum AVSampleFormat {
AV_SAMPLE_FMT_NONE = -1,
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NB
};
int sampleSize = mAudioFormat.sampleSize();
QAudioFormat::SampleType sampleType = mAudioFormat.sampleType();
switch (sampleType) {
case QAudioFormat::Unknown:
return AV_SAMPLE_FMT_NONE;
case QAudioFormat::SignedInt:
if (sampleSize == 16)
return AV_SAMPLE_FMT_S16;
if (sampleSize == 32)
return AV_SAMPLE_FMT_S32;
break;
case QAudioFormat::UnSignedInt:
if (sampleSize == 8)
return AV_SAMPLE_FMT_U8;
break;
case QAudioFormat::Float:
return AV_SAMPLE_FMT_FLT;
default:
return AV_SAMPLE_FMT_NONE;
}
return AV_SAMPLE_FMT_NONE;
}
QString UBMicrophoneInput::codec()
{
return mAudioFormat.codec();
}
qint64 UBMicrophoneInput::processUSecs() const
{
return mAudioInput->processedUSecs();
}
bool UBMicrophoneInput::init() bool UBMicrophoneInput::init()
{ {
if (mAudioDeviceInfo.isNull()) { if (mAudioDeviceInfo.isNull()) {
@ -94,15 +41,14 @@ bool UBMicrophoneInput::init()
mAudioDeviceInfo = QAudioDeviceInfo::defaultInputDevice(); mAudioDeviceInfo = QAudioDeviceInfo::defaultInputDevice();
} }
qDebug() << "Input device name: " << mAudioDeviceInfo.deviceName();
mAudioFormat = mAudioDeviceInfo.preferredFormat(); mAudioFormat = mAudioDeviceInfo.preferredFormat();
mAudioInput = new QAudioInput(mAudioDeviceInfo, mAudioFormat, NULL); mAudioInput = new QAudioInput(mAudioDeviceInfo, mAudioFormat, NULL);
connect(mAudioInput, SIGNAL(stateChanged(QAudio::State)), connect(mAudioInput, SIGNAL(stateChanged(QAudio::State)),
this, SLOT(onAudioInputStateChanged(QAudio::State))); this, SLOT(onAudioInputStateChanged(QAudio::State)));
qDebug() << "Input device name: " << mAudioDeviceInfo.deviceName();
qDebug() << "Input sample format: " << mAudioFormat.sampleSize() << "bit" qDebug() << "Input sample format: " << mAudioFormat.sampleSize() << "bit"
<< mAudioFormat.sampleType() << "at" << mAudioFormat.sampleRate() << "Hz" << mAudioFormat.sampleType() << "at" << mAudioFormat.sampleRate() << "Hz"
<< "; codec: " << mAudioFormat.codec(); << "; codec: " << mAudioFormat.codec();
@ -112,8 +58,6 @@ bool UBMicrophoneInput::init()
void UBMicrophoneInput::start() void UBMicrophoneInput::start()
{ {
qDebug() << "starting audio input";
mIODevice = mAudioInput->start(); mIODevice = mAudioInput->start();
connect(mIODevice, SIGNAL(readyRead()), connect(mIODevice, SIGNAL(readyRead()),
@ -165,6 +109,74 @@ void UBMicrophoneInput::setInputDevice(QString name)
} }
int UBMicrophoneInput::channelCount()
{
return mAudioFormat.channelCount();
}
int UBMicrophoneInput::sampleRate()
{
return mAudioFormat.sampleRate();
}
/* Return the sample size in bits */
int UBMicrophoneInput::sampleSize()
{
return mAudioFormat.sampleSize();
}
/** Return the sample format in FFMpeg style (AVSampleFormat enum) */
int UBMicrophoneInput::sampleFormat()
{
enum AVSampleFormat {
AV_SAMPLE_FMT_NONE = -1,
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NB
};
int sampleSize = mAudioFormat.sampleSize();
QAudioFormat::SampleType sampleType = mAudioFormat.sampleType();
switch (sampleType) {
case QAudioFormat::Unknown:
return AV_SAMPLE_FMT_NONE;
case QAudioFormat::SignedInt:
if (sampleSize == 16)
return AV_SAMPLE_FMT_S16;
if (sampleSize == 32)
return AV_SAMPLE_FMT_S32;
break;
case QAudioFormat::UnSignedInt:
if (sampleSize == 8)
return AV_SAMPLE_FMT_U8;
break;
case QAudioFormat::Float:
return AV_SAMPLE_FMT_FLT;
default:
return AV_SAMPLE_FMT_NONE;
}
return AV_SAMPLE_FMT_NONE;
}
QString UBMicrophoneInput::codec()
{
return mAudioFormat.codec();
}
static qint64 uSecsElapsed = 0; static qint64 uSecsElapsed = 0;
void UBMicrophoneInput::onDataReady() void UBMicrophoneInput::onDataReady()
{ {
@ -189,7 +201,6 @@ void UBMicrophoneInput::onDataReady()
void UBMicrophoneInput::onAudioInputStateChanged(QAudio::State state) void UBMicrophoneInput::onAudioInputStateChanged(QAudio::State state)
{ {
qDebug() << "Audio input state changed to " << state;
switch (state) { switch (state) {
case QAudio::StoppedState: case QAudio::StoppedState:
if (mAudioInput->error() != QAudio::NoError) { if (mAudioInput->error() != QAudio::NoError) {
@ -227,8 +238,7 @@ quint8 UBMicrophoneInput::audioLevel(const QByteArray &data)
double rms = sqrt(sum/n_samples); double rms = sqrt(sum/n_samples);
// The vu meter looks a bit better when the RMS isn't displayed linearly, as perceived sound // The vu meter looks a bit better when the RMS isn't displayed linearly, as perceived sound
// level increases logarithmically. So here RMS can be substituted by something like // level increases logarithmically. So here RMS is substituted by rms^(1/e)
// rms^(1/e)
rms = pow(rms, 1./exp(1)); rms = pow(rms, 1./exp(1));
return UINT8_MAX * rms; return UINT8_MAX * rms;
@ -237,7 +247,8 @@ quint8 UBMicrophoneInput::audioLevel(const QByteArray &data)
/** /**
* @brief Calculate one sample's level relative to its maximum value * @brief Calculate one sample's level relative to its maximum value
* @param sample One sample, in the format specified by mAudioFormat * @param sample One sample, in the format specified by mAudioFormat
* @return A double between 0 and 1.0, where 1.0 is the maximum value the sample can take. * @return A double between 0 and 1.0, where 1.0 is the maximum value the sample can take,
* or -1 if the value couldn't be calculated.
*/ */
double UBMicrophoneInput::sampleRelativeLevel(const char* sample) double UBMicrophoneInput::sampleRelativeLevel(const char* sample)
{ {

@ -1,3 +1,24 @@
/*
* Copyright (C) 2015-2016 Département de l'Instruction Publique (DIP-SEM)
*
* This file is part of OpenBoard.
*
* OpenBoard is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, version 3 of the License,
* with a specific linking exception for the OpenSSL project's
* "OpenSSL" library (or with modified versions of it that use the
* same license as the "OpenSSL" library).
*
* OpenBoard is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with OpenBoard. If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef UBMICROPHONEINPUT_H #ifndef UBMICROPHONEINPUT_H
#define UBMICROPHONEINPUT_H #define UBMICROPHONEINPUT_H
@ -5,7 +26,9 @@
#include <QAudioInput> #include <QAudioInput>
/** /**
* @brief The UBMicrophoneInput class captures uncompressed sound from a microphone * @brief The UBMicrophoneInput class captures uncompressed sound from a microphone.
*
* Audio samples can be read by connecting to the dataAvailable signal.
*/ */
class UBMicrophoneInput : public QObject class UBMicrophoneInput : public QObject
{ {
@ -28,10 +51,8 @@ public:
int sampleFormat(); int sampleFormat();
QString codec(); QString codec();
qint64 processUSecs() const;
signals: signals:
/// Send the new volume, between 0 and 255 /// Send the new audio level, between 0 and 255
void audioLevelChanged(quint8 level); void audioLevelChanged(quint8 level);
/// Emitted when new audio data is available /// Emitted when new audio data is available

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