Fix BWE bug where audio has timestamps in us.

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
new file mode 100644
index 0000000000000000000000000000000000000000..cf5314cea10081c878bfcb37268605f4e24d7c22
--- /dev/null
+++ b/webrtc/video/audio_receive_stream_unittest.cc
@@ -0,0 +1,77 @@
+/*
+ *  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