Merge branch 'dev-ffmpeg-osx' into dev

OpenBoard.pro

@@ -44,6 +44,8 @@ QT += printsupport
 QT += core
 
 INCLUDEPATH += src
+INCLUDEPATH += /usr/local/opt/openssl/include
+INCLUDEPATH += /usr/local/opt/ffmpeg/include
 
 include($$THIRD_PARTY_PATH/libs.pri)
 include(src/adaptors/adaptors.pri)
@@ -162,6 +164,8 @@ macx {
    LIBS += -framework AVFoundation
    LIBS += -framework CoreMedia
    LIBS += -lcrypto
+   LIBS += -L/usr/local/opt/openssl/lib
+   LIBS += -L/usr/local/opt/ffmpeg/lib
 
    CONFIG(release, debug|release):CONFIG += x86_64
    CONFIG(debug, debug|release):CONFIG += x86_64

src/podcast/UBPodcastController.cpp

@@ -62,8 +62,8 @@
     #include "windowsmedia/UBWindowsMediaVideoEncoder.h"
     #include "windowsmedia/UBWaveRecorder.h"
 #elif defined(Q_OS_OSX)
-    #include "quicktime/UBQuickTimeVideoEncoder.h"
-    #include "quicktime/UBAudioQueueRecorder.h"
+    #include "ffmpeg/UBFFmpegVideoEncoder.h"
+    #include "ffmpeg/UBMicrophoneInput.h"
 #elif defined(Q_OS_LINUX)
     #include "ffmpeg/UBFFmpegVideoEncoder.h"
     #include "ffmpeg/UBMicrophoneInput.h"
@@ -311,7 +311,7 @@ void UBPodcastController::start()
 #ifdef Q_OS_WIN
         mVideoEncoder = new UBWindowsMediaVideoEncoder(this);  //deleted on stop
 #elif defined(Q_OS_OSX)
-        mVideoEncoder = new UBQuickTimeVideoEncoder(this);  //deleted on stop
+        mVideoEncoder = new UBFFmpegVideoEncoder(this);
 #elif defined(Q_OS_LINUX)
         mVideoEncoder = new UBFFmpegVideoEncoder(this);
 #endif
@@ -808,7 +808,7 @@ QStringList UBPodcastController::audioRecordingDevices()
 #ifdef Q_OS_WIN
     devices = UBWaveRecorder::waveInDevices();
 #elif defined(Q_OS_OSX)
-    devices = UBAudioQueueRecorder::waveInDevices();
+    devices = UBMicrophoneInput::availableDevicesNames();
 #elif defined(Q_OS_LINUX)
     devices = UBMicrophoneInput::availableDevicesNames();
 #endif

src/podcast/ffmpeg/UBFFmpegVideoEncoder.cpp

@@ -369,6 +369,8 @@ bool UBFFmpegVideoEncoder::init()
         connect(mAudioInput, SIGNAL(dataAvailable(QByteArray)),
                 this, SLOT(onAudioAvailable(QByteArray)));
 
+        mAudioInput->setInputDevice(audioRecordingDevice());
+
         if (!mAudioInput->init()) {
             setLastErrorMessage("Couldn't initialize audio input");
             return false;
@@ -392,13 +394,20 @@ bool UBFFmpegVideoEncoder::init()
         c = mAudioStream->codec;
 
         c->bit_rate = 96000;
-        c->sample_fmt = audioCodec->sample_fmts[0]; // FLTP by default for AAC
+        c->sample_fmt  = audioCodec->sample_fmts ? audioCodec->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;// FLTP by default for AAC
         c->sample_rate = mAudioSampleRate;
-        c->channels = 2;
-        c->channel_layout = av_get_default_channel_layout(c->channels);
-        c->profile = FF_PROFILE_AAC_MAIN;
-        c->time_base = {1, mAudioSampleRate};
-        c->strict_std_compliance = -2; // Enable use of experimental codec
+        c->channel_layout = AV_CH_LAYOUT_STEREO;
+        c->channels  = av_get_channel_layout_nb_channels(c->channel_layout);
+
+        //deprecated on ffmpeg 4
+        c->strict_std_compliance = -2;// Enable use of experimental codec
+
+        //https://trac.ffmpeg.org/wiki/Encode/H.264#Profile
+        //Omit this unless your target device only supports a certain profile
+        //(see https://trac.ffmpeg.org/wiki/Encode/H.264#Compatibility).
+        //c->profile = FF_PROFILE_AAC_MAIN;
+
+        c->time_base = { 1, c->sample_rate };
 
         if (mOutputFormatContext->oformat->flags & AVFMT_GLOBALHEADER)
             c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
@@ -562,9 +571,11 @@ void UBFFmpegVideoEncoder::processAudio(QByteArray &data)
     }
 
     // Convert to destination format
+
     ret = swr_convert(mSwrContext,
                       outSamples, outSamplesCount,
                       (const uint8_t **)&inSamples, inSamplesCount);
+
     if (ret < 0) {
         qWarning() << "Error converting audio samples: " << avErrorToQString(ret);
         return;

src/podcast/ffmpeg/UBFFmpegVideoEncoder.h

@@ -36,7 +36,10 @@ extern "C" {
     #include <libswscale/swscale.h>
 
 // Due to the whole ffmpeg / libAV silliness, we have to support libavresample on some platforms
-#if LIBAVFORMAT_VERSION_MICRO > 100
+// Should libswresample be used with libavformat.*.*.100 ? compiles fine but crashes in swr_convert (internal call of soxr_create causes a SIGSEV (EXC_BAD_ACCESS))
+// Bad config ?
+// With libavresample, no crash, but the sound is inaudible (lot of noise)
+#if LIBAVFORMAT_VERSION_MICRO >= 100
     #include <libswresample/swresample.h>
 #else
     #include <libavresample/avresample.h>

src/podcast/ffmpeg/UBMicrophoneInput.cpp

@@ -42,6 +42,18 @@ bool UBMicrophoneInput::init()
     }
 
     mAudioFormat = mAudioDeviceInfo.preferredFormat();
+
+    /*
+     *  https://ffmpeg.org/doxygen/3.1/group__lavu__sampfmts.html#gaf9a51ca15301871723577c730b5865c5
+        The data described by the sample format is always in native-endian order.
+        Sample values can be expressed by native C types, hence the lack of a signed 24-bit sample format
+        even though it is a common raw audio data format.
+
+        If a signed 24-bit sample format is natively preferred, we a set signed 16-bit sample format instead.
+    */
+    if (mAudioFormat.sampleSize() == 24)
+        mAudioFormat.setSampleSize(16);
+
     mAudioInput = new QAudioInput(mAudioDeviceInfo, mAudioFormat, NULL);
 
     connect(mAudioInput, SIGNAL(stateChanged(QAudio::State)),
@@ -294,6 +306,8 @@ QString UBMicrophoneInput::getErrorString(QAudio::Error errorCode)
         case QAudio::FatalError :
             return "Fatal error; audio device unusable";
 
+        default:
+            return "unhandled error...";
     }
     return "";
 }

src/podcast/podcast.pri

@@ -23,15 +23,33 @@ win32 {
 }
 
 macx {
+    CONFIG += c++11
 
-    SOURCES  += src/podcast/quicktime/UBQuickTimeVideoEncoder.cpp \
-                src/podcast/quicktime/UBAudioQueueRecorder.cpp
+    SOURCES  += src/podcast/ffmpeg/UBFFmpegVideoEncoder.cpp \
+                src/podcast/ffmpeg/UBMicrophoneInput.cpp
 
-    HEADERS  += src/podcast/quicktime/UBQuickTimeVideoEncoder.h \
-                src/podcast/quicktime/UBQuickTimeFile.h \
-                src/podcast/quicktime/UBAudioQueueRecorder.h
+    HEADERS  += src/podcast/ffmpeg/UBFFmpegVideoEncoder.h \
+                src/podcast/ffmpeg/UBMicrophoneInput.h
 
-    OBJECTIVE_SOURCES += src/podcast/quicktime/UBQuickTimeFile.mm
+    LIBS += -lavformat -lavcodec -lswscale  -lswresample -lavutil \
+        -lpthread -lvpx -lvorbisenc -lfreetype -llzma -lbz2 -lz -ldl -lavutil -lm
+
+    # (ffmpeg-4.0 with all options (to clean))
+    # brew install ffmpeg --with-chromaprint --with-fdk-aac --with-libass --with-librsvg --with-libsoxr --with-libssh --with-tesseract
+    #--with-libvidstab --with-opencore-amr --with-openh264 --with-openjpeg --with-openssl --with-rtmpdump --with-rubberband --with-sdl2
+    #--with-snappy --with-tools --with-webp --with-x265 --with-xz --with-zeromq --with-zimg
+    # brew install opus
+
+    LIBS += -L/usr/local/opt/x264/lib
+    LIBS += -L/usr/local/opt/sdl/lib
+    LIBS += -L/usr/local/opt/libvorbis/lib
+    LIBS += -L/usr/local/opt/libvpx/lib
+    LIBS += -L/usr/local/opt/theora/lib
+    LIBS += -L/usr/local/opt/libogg/lib
+    LIBS += -L/usr/local/opt/opus/lib
+    LIBS += -L/usr/local/opt/lame/lib
+    LIBS += -L/usr/local/opt/fdk-aac/lib
+    LIBS += -L/usr/local/opt/libass/lib
 }
 
 linux-g++* {