Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 43 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.h
new file mode 100644
index 0000000000..be1705cae1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+
+#include <bitset>
+#include <memory>
+
+#include "modules/audio_coding/neteq/tools/packet.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for an object delivering RTP packets to test applications.
+class PacketSource {
+ public:
+  PacketSource();
+  virtual ~PacketSource();
+
+  PacketSource(const PacketSource&) = delete;
+  PacketSource& operator=(const PacketSource&) = delete;
+
+  // Returns next packet. Returns nullptr if the source is depleted, or if an
+  // error occurred.
+  virtual std::unique_ptr<Packet> NextPacket() = 0;
+
+  virtual void FilterOutPayloadType(uint8_t payload_type);
+
+ protected:
+  std::bitset<128> filter_;  // Payload type is 7 bits in the RFC.
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_