From bd48a2feb5d134833f080b1d093b5c833c59a9be Mon Sep 17 00:00:00 2001
From: f4exb <f4exb06@gmail.com>
Date: Tue, 19 Feb 2019 02:07:26 +0100
Subject: [PATCH] Copy audio to UDP/RTP: Opus implementation (5)

---
 sdrbase/audio/audionetsink.cpp | 2 ++
 sdrbase/audio/audioopus.cpp    | 7 ++++++-
 sdrbase/audio/audioopus.h      | 2 ++
 sdrgui/audio.md                | 7 ++++---
 sdrgui/gui/audiodialog.cpp     | 2 ++
 5 files changed, 16 insertions(+), 4 deletions(-)

diff --git a/sdrbase/audio/audionetsink.cpp b/sdrbase/audio/audionetsink.cpp
index 275539c81..5b3315f59 100644
--- a/sdrbase/audio/audionetsink.cpp
+++ b/sdrbase/audio/audionetsink.cpp
@@ -175,6 +175,8 @@ void AudioNetSink::setNewCodecData()
             << " Fs: " << m_sampleRate/m_decimation
             << " stereo: " << m_stereo;
         m_opus.setEncoder(m_sampleRate/m_decimation, m_stereo ? 2 : 1);
+        m_codecInputIndex = 0;
+        m_bufferIndex = 0;
     }
 
     setDecimationFilters();
diff --git a/sdrbase/audio/audioopus.cpp b/sdrbase/audio/audioopus.cpp
index cdf5b2057..1a964356f 100644
--- a/sdrbase/audio/audioopus.cpp
+++ b/sdrbase/audio/audioopus.cpp
@@ -20,12 +20,14 @@
 
 #include "opus/opus.h"
 #include <QDebug>
+#include <QMutexLocker>
 
 #include "audioopus.h"
 
 AudioOpus::AudioOpus() :
     m_encoderState(0),
-    m_encoderOK(false)
+    m_encoderOK(false),
+    m_mutex(QMutex::Recursive)
 {
     qDebug("AudioOpus::AudioOpus: libopus version %s", opus_get_version_string());
 }
@@ -41,6 +43,7 @@ void AudioOpus::setEncoder(int32_t fs, int nChannels)
 {
     int error;
     bool newInstance = true;
+    QMutexLocker mutexLocker(&m_mutex);
 
     if (m_encoderState)
     {
@@ -85,6 +88,8 @@ void AudioOpus::setEncoder(int32_t fs, int nChannels)
 
 int AudioOpus::encode(int frameSize, int16_t *in, uint8_t *out)
 {
+    QMutexLocker mutexLocker(&m_mutex);
+
     int nbBytes = opus_encode(m_encoderState, in, frameSize, out, m_maxPacketSize);
 
     if (nbBytes < 0)
diff --git a/sdrbase/audio/audioopus.h b/sdrbase/audio/audioopus.h
index 14fdee121..b9aba5e06 100644
--- a/sdrbase/audio/audioopus.h
+++ b/sdrbase/audio/audioopus.h
@@ -19,6 +19,7 @@
 #define SDRBASE_AUDIO_AUDIOOPUS_H_
 
 #include <stdint.h>
+#include <QMutex>
 #include "export.h"
 
 class OpusEncoder;
@@ -38,6 +39,7 @@ public:
 private:
     OpusEncoder *m_encoderState;
     bool m_encoderOK;
+    QMutex m_mutex;
 };
 
 #endif /* SDRBASE_AUDIO_AUDIOOPUS_H_ */
diff --git a/sdrgui/audio.md b/sdrgui/audio.md
index 9cb03a137..3c158acde 100644
--- a/sdrgui/audio.md
+++ b/sdrgui/audio.md
@@ -129,12 +129,13 @@ m=audio 9998 RTP/AVP 9
 a=rtpmap:9 G722/8000/1
 ```
 
-For Opus mono:
+For Opus mono or stereo:
 
 ```
 c=IN IP4 192.168.0.34
-m=audio 9998 RTP/AVP 101
-a=rtpmap:101 opus/48000/1
+m=audio 9998 RTP/AVP 96
+a=rtpmap:96 opus/48000/2
+a=fmtp:96 cbr=1
 ```
 
 &#9758; Note that on Android clients VLC has trouble working with the RTP stream (choppy audio, hanging unexpectedly...) therefore [MX player](https://play.google.com/store/apps/details?id=com.mxtech.videoplayer.ad&hl=en) is recommended.
diff --git a/sdrgui/gui/audiodialog.cpp b/sdrgui/gui/audiodialog.cpp
index ca1f6514a..665cb5429 100644
--- a/sdrgui/gui/audiodialog.cpp
+++ b/sdrgui/gui/audiodialog.cpp
@@ -317,6 +317,8 @@ void AudioDialogX::updateOutputSDPString()
         break;
     case AudioOutput::UDPCodecOpus:
         format = "opus";
+        nChannels = 2; // always 2 even for mono
+        effectiveSampleRate = 48000; // always 48000 regardless of input rate
         break;
     case AudioOutput::UDPCodecL16:
     default: