Fix BWE bug where audio has timestamps in us.
The BWE expects arrival timestamps in ms, while the audio path delivered
them in us, causing the BWE to break down under the combined audio/video
BWE experiment. This was introduced in r9892 (https://chromium.googlesource.com/external/webrtc/+/68786d20400f1f3744ad83549325665c18ea9e5b).
BUG=webrtc:4758
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1360913004 .
Cr-Commit-Position: refs/heads/master@{#10032}
diff --git a/webrtc/video/audio_receive_stream.cc b/webrtc/video/audio_receive_stream.cc
index b8da1bb..a1cf2ca 100644
--- a/webrtc/video/audio_receive_stream.cc
+++ b/webrtc/video/audio_receive_stream.cc
@@ -102,7 +102,7 @@
header.extension.hasAbsoluteSendTime) {
int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
if (packet_time.timestamp >= 0)
- arrival_time_ms = packet_time.timestamp;
+ arrival_time_ms = (packet_time.timestamp + 500) / 1000;
size_t payload_size = length - header.headerLength;
remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
header, false);
diff --git a/webrtc/video/audio_receive_stream_unittest.cc b/webrtc/video/audio_receive_stream_unittest.cc
new file mode 100644
index 0000000..cf5314c
--- /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
diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi
index 2515ebd..06bb25f 100644
--- a/webrtc/webrtc_tests.gypi
+++ b/webrtc/webrtc_tests.gypi
@@ -154,6 +154,7 @@
'test/common_unittest.cc',
'test/testsupport/metrics/video_metrics_unittest.cc',
'tools/agc/agc_manager_unittest.cc',
+ 'video/audio_receive_stream_unittest.cc',
'video/bitrate_estimator_tests.cc',
'video/end_to_end_tests.cc',
'video/packet_injection_tests.cc',
