fixed some issues on audio resampling

src/podcast/ffmpeg/UBFFmpegVideoEncoder.cpp

@@ -392,13 +392,17 @@ 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);
+
+        //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;
@@ -562,11 +566,6 @@ void UBFFmpegVideoEncoder::processAudio(QByteArray &data)
     }
 
     // Convert to destination format
-    qDebug() << mSwrContext;
-    qDebug() << outSamples;
-    qDebug() << outSamplesCount;
-    qDebug() << inSamples;
-    qDebug() << inSamplesCount;
 
     ret = swr_convert(mSwrContext,
                       outSamples, outSamplesCount,

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

@@ -31,8 +31,8 @@ macx {
     HEADERS  += src/podcast/ffmpeg/UBFFmpegVideoEncoder.h \
                 src/podcast/ffmpeg/UBMicrophoneInput.h
 
-    LIBS += -lavformat -lavcodec -lswscale -lavutil \
-    -lpthread -lvpx -lvorbisenc -lfreetype -llzma -lbz2 -lz -ldl -lswresample -lswscale -lavutil -lm
+    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