1 files changed, 53 insertions, 0 deletions
diff --git a/host/frontend/gcastv2/webrtc/include/webrtc/G711Packetizer.h b/host/frontend/gcastv2/webrtc/include/webrtc/G711Packetizer.h
new file mode 100644
index 000000000..82fcb4c07
--- /dev/null
+++ b/host/frontend/gcastv2/webrtc/include/webrtc/G711Packetizer.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include "Packetizer.h"
+
+#include <https/RunLoop.h>
+
+#include <memory>
+
+#include <source/StreamingSource.h>
+
+struct G711Packetizer : public Packetizer {
+
+    using StreamingSource = android::StreamingSource;
+
+    enum class Mode {
+        ALAW,
+        ULAW
+    };
+    explicit G711Packetizer(
+            Mode mode,
+            std::shared_ptr<RunLoop> runLoop,
+            std::shared_ptr<StreamingSource> audioSource);
+
+    uint32_t rtpNow() const override;
+
+private:
+    using SBuffer = android::SBuffer;
+
+    Mode mMode;
+    std::shared_ptr<RunLoop> mRunLoop;
+
+    bool mFirstInTalkspurt;
+
+    void packetize(const std::shared_ptr<SBuffer> &accessUnit, int64_t timeUs);
+};
+
+