Split webrtc/video into webrtc/{audio,call,video}.

webrtc/video/audio_receive_stream_unittest.cc

Index: webrtc/video/audio_receive_stream_unittest.cc
diff --git a/webrtc/video/audio_receive_stream_unittest.cc b/webrtc/video/audio_receive_stream_unittest.cc
deleted file mode 100644
index cf5314cea10081c878bfcb37268605f4e24d7c22..0000000000000000000000000000000000000000
--- a/webrtc/video/audio_receive_stream_unittest.cc
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- *  Copyright (c) 2015 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.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-
-#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/video/audio_receive_stream.h"
-
-namespace webrtc {
-
-const size_t kAbsoluteSendTimeLength = 4;
-
-void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
-                                    int id,
-                                    uint32_t abs_send_time) {
-  const size_t kRtpOneByteHeaderLength = 4;
-  const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
-  ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId);
-
-  const uint32_t kPosLength = 2;
-  ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength,
-                                       kAbsoluteSendTimeLength / 4);
-
-  const uint8_t kLengthOfData = 3;
-  buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1);
-  ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian(
-      buffer + kRtpOneByteHeaderLength + 1, abs_send_time);
-}
-
-size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
-                                      int extension_id,
-                                      uint32_t abs_send_time) {
-  header[0] = 0x80;   // Version 2.
-  header[0] |= 0x10;  // Set extension bit.
-  header[1] = 100;    // Payload type.
-  header[1] |= 0x80;  // Marker bit is set.
-  ByteWriter<uint16_t>::WriteBigEndian(header + 2, 0x1234);  // Sequence number.
-  ByteWriter<uint32_t>::WriteBigEndian(header + 4, 0x5678);  // Timestamp.
-  ByteWriter<uint32_t>::WriteBigEndian(header + 8, 0x4321);  // SSRC.
-  int32_t rtp_header_length = kRtpHeaderSize;
-
-  BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
-                                 abs_send_time);
-  rtp_header_length += kAbsoluteSendTimeLength;
-  return rtp_header_length;
-}
-
-TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
-  MockRemoteBitrateEstimator rbe;
-  AudioReceiveStream::Config config;
-  config.combined_audio_video_bwe = true;
-  config.voe_channel_id = 1;
-  const int kAbsSendTimeId = 3;
-  config.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
-  internal::AudioReceiveStream recv_stream(&rbe, config);
-  uint8_t rtp_packet[30];
-  const int kAbsSendTimeValue = 1234;
-  CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
-  PacketTime packet_time(5678000, 0);
-  const size_t kExpectedHeaderLength = 20;
-  EXPECT_CALL(rbe, IncomingPacket(packet_time.timestamp / 1000,
-                                  sizeof(rtp_packet) - kExpectedHeaderLength,
-                                  testing::_, false))
-      .Times(1);
-  EXPECT_TRUE(
-      recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
-}
-}  // namespace webrtc