Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 06:15:48 +02:00
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
--- a/modules/audio_coding/neteq/mock/mock_packet_buffer.h
+++ b/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+
+#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockPacketBuffer : public PacketBuffer {
+ public:
+  MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
+      : PacketBuffer(max_number_of_packets, tick_timer) {}
+  virtual ~MockPacketBuffer() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD0(Flush,
+      void());
+  MOCK_CONST_METHOD0(Empty,
+      bool());
+  int InsertPacket(Packet&& packet, StatisticsCalculator* stats) {
+    return InsertPacketWrapped(&packet, stats);
+  }
+  // Since gtest does not properly support move-only types, InsertPacket is
+  // implemented as a wrapper. You'll have to implement InsertPacketWrapped
+  // instead and move from |*packet|.
+  MOCK_METHOD2(InsertPacketWrapped,
+               int(Packet* packet, StatisticsCalculator* stats));
+  MOCK_METHOD5(InsertPacketList,
+               int(PacketList* packet_list,
+                   const DecoderDatabase& decoder_database,
+                   rtc::Optional<uint8_t>* current_rtp_payload_type,
+                   rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
+                   StatisticsCalculator* stats));
+  MOCK_CONST_METHOD1(NextTimestamp,
+      int(uint32_t* next_timestamp));
+  MOCK_CONST_METHOD2(NextHigherTimestamp,
+      int(uint32_t timestamp, uint32_t* next_timestamp));
+  MOCK_CONST_METHOD0(PeekNextPacket,
+      const Packet*());
+  MOCK_METHOD0(GetNextPacket,
+      rtc::Optional<Packet>());
+  MOCK_METHOD1(DiscardNextPacket, int(StatisticsCalculator* stats));
+  MOCK_METHOD3(DiscardOldPackets,
+               void(uint32_t timestamp_limit,
+                    uint32_t horizon_samples,
+                    StatisticsCalculator* stats));
+  MOCK_METHOD2(DiscardAllOldPackets,
+               void(uint32_t timestamp_limit, StatisticsCalculator* stats));
+  MOCK_CONST_METHOD0(NumPacketsInBuffer,
+      size_t());
+  MOCK_METHOD1(IncrementWaitingTimes,
+      void(int));
+  MOCK_CONST_METHOD0(current_memory_bytes,
+      int());
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_