Wire-up BWE feedback for audio receive streams.
Also wires up receiving transport sequence numbers.
BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1535963002 .
Cr-Commit-Position: refs/heads/master@{#11220}
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 01e8a2b..eb008b3 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -14,10 +14,16 @@
#include "webrtc/audio/audio_receive_stream.h"
#include "webrtc/audio/conversion.h"
+#include "webrtc/call/mock/mock_congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
+#include "webrtc/modules/pacing/packet_router.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/modules/utility/include/mock/mock_process_thread.h"
+#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
#include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/video/call_stats.h"
namespace webrtc {
namespace test {
@@ -40,9 +46,11 @@
const int kChannelId = 2;
const uint32_t kRemoteSsrc = 1234;
const uint32_t kLocalSsrc = 5678;
-const size_t kAbsoluteSendTimeLength = 4;
+const size_t kOneByteExtensionHeaderLength = 4;
+const size_t kOneByteExtensionLength = 4;
const int kAbsSendTimeId = 2;
const int kAudioLevelId = 3;
+const int kTransportSequenceNumberId = 4;
const int kJitterBufferDelay = -7;
const int kPlayoutBufferDelay = 302;
const unsigned int kSpeechOutputLevel = 99;
@@ -55,7 +63,12 @@
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
- ConfigHelper() {
+ ConfigHelper()
+ : simulated_clock_(123456),
+ call_stats_(&simulated_clock_),
+ congestion_controller_(&process_thread_,
+ &call_stats_,
+ &bitrate_observer_) {
using testing::Invoke;
EXPECT_CALL(voice_engine_,
@@ -77,6 +90,14 @@
EXPECT_CALL(*channel_proxy_,
SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
+ EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(
+ nullptr, nullptr, &packet_router_))
+ .Times(1);
+ EXPECT_CALL(congestion_controller_, packet_router())
+ .WillOnce(Return(&packet_router_));
+ EXPECT_CALL(*channel_proxy_,
+ SetCongestionControlObjects(nullptr, nullptr, nullptr))
+ .Times(1);
return channel_proxy_;
}));
stream_config_.voe_channel_id = kChannelId;
@@ -88,6 +109,9 @@
RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
}
+ MockCongestionController* congestion_controller() {
+ return &congestion_controller_;
+ }
MockRemoteBitrateEstimator* remote_bitrate_estimator() {
return &remote_bitrate_estimator_;
}
@@ -95,11 +119,19 @@
rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
MockVoiceEngine& voice_engine() { return voice_engine_; }
+ void SetupMockForBweFeedback(bool send_side_bwe) {
+ EXPECT_CALL(congestion_controller_,
+ GetRemoteBitrateEstimator(send_side_bwe))
+ .WillOnce(Return(&remote_bitrate_estimator_));
+ EXPECT_CALL(remote_bitrate_estimator_,
+ RemoveStream(stream_config_.rtp.remote_ssrc));
+ }
+
void SetupMockForGetStats() {
using testing::DoAll;
using testing::SetArgReferee;
- EXPECT_TRUE(channel_proxy_);
+ ASSERT_TRUE(channel_proxy_);
EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
.WillOnce(Return(kCallStats));
EXPECT_CALL(*channel_proxy_, GetDelayEstimate())
@@ -116,6 +148,12 @@
}
private:
+ SimulatedClock simulated_clock_;
+ CallStats call_stats_;
+ PacketRouter packet_router_;
+ testing::NiceMock<MockBitrateObserver> bitrate_observer_;
+ testing::NiceMock<MockProcessThread> process_thread_;
+ MockCongestionController congestion_controller_;
MockRemoteBitrateEstimator remote_bitrate_estimator_;
testing::StrictMock<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
@@ -123,39 +161,43 @@
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
};
-void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
- int id,
- uint32_t abs_send_time) {
- const size_t kRtpOneByteHeaderLength = 4;
+void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
+ int id,
+ uint32_t extension_value,
+ size_t value_length) {
const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
- ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId);
+ ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId);
+ it += 2;
- 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);
+ ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4);
+ it += 2;
+ const size_t kExtensionDataLength = kOneByteExtensionLength - 1;
+ uint32_t shifted_value = extension_value
+ << (8 * (kExtensionDataLength - value_length));
+ *it = (id << 4) + (value_length - 1);
+ ++it;
+ ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it),
+ shifted_value);
}
-size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
- int extension_id,
- uint32_t abs_send_time) {
+std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
+ int extension_id,
+ uint32_t extension_value,
+ size_t value_length) {
+ std::vector<uint8_t> header;
+ header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength +
+ kOneByteExtensionLength);
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 = webrtc::kRtpHeaderSize;
+ 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.
- BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
- abs_send_time);
- rtp_header_length += kAbsoluteSendTimeLength;
- return rtp_header_length;
+ BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id,
+ extension_value, value_length);
+ return header;
}
} // namespace
@@ -178,32 +220,73 @@
TEST(AudioReceiveStreamTest, ConstructDestruct) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
- helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
+ helper.congestion_controller(), helper.config(), helper.audio_state());
+}
+
+MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
+ return arg.extension.hasAbsoluteSendTime ==
+ expected_extension.hasAbsoluteSendTime &&
+ arg.extension.absoluteSendTime ==
+ expected_extension.absoluteSendTime &&
+ arg.extension.hasTransportSequenceNumber ==
+ expected_extension.hasTransportSequenceNumber &&
+ arg.extension.transportSequenceNumber ==
+ expected_extension.transportSequenceNumber;
}
TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
ConfigHelper helper;
helper.config().combined_audio_video_bwe = true;
+ helper.SetupMockForBweFeedback(false);
internal::AudioReceiveStream recv_stream(
- helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
- uint8_t rtp_packet[30];
+ helper.congestion_controller(), helper.config(), helper.audio_state());
const int kAbsSendTimeValue = 1234;
- CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
+ std::vector<uint8_t> rtp_packet =
+ CreateRtpHeaderWithOneByteExtension(kAbsSendTimeId, kAbsSendTimeValue, 3);
PacketTime packet_time(5678000, 0);
const size_t kExpectedHeaderLength = 20;
+ RTPHeaderExtension expected_extension;
+ expected_extension.hasAbsoluteSendTime = true;
+ expected_extension.absoluteSendTime = kAbsSendTimeValue;
EXPECT_CALL(*helper.remote_bitrate_estimator(),
IncomingPacket(packet_time.timestamp / 1000,
- sizeof(rtp_packet) - kExpectedHeaderLength,
- testing::_, false))
+ rtp_packet.size() - kExpectedHeaderLength,
+ VerifyHeaderExtension(expected_extension), false))
.Times(1);
EXPECT_TRUE(
- recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
+ recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
+}
+
+TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
+ ConfigHelper helper;
+ helper.config().combined_audio_video_bwe = true;
+ helper.config().rtp.transport_cc = true;
+ helper.config().rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
+ helper.SetupMockForBweFeedback(true);
+ internal::AudioReceiveStream recv_stream(
+ helper.congestion_controller(), helper.config(), helper.audio_state());
+ const int kTransportSequenceNumberValue = 1234;
+ std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
+ kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
+ PacketTime packet_time(5678000, 0);
+ const size_t kExpectedHeaderLength = 20;
+ RTPHeaderExtension expected_extension;
+ expected_extension.hasTransportSequenceNumber = true;
+ expected_extension.transportSequenceNumber = kTransportSequenceNumberValue;
+ EXPECT_CALL(*helper.remote_bitrate_estimator(),
+ IncomingPacket(packet_time.timestamp / 1000,
+ rtp_packet.size() - kExpectedHeaderLength,
+ VerifyHeaderExtension(expected_extension), false))
+ .Times(1);
+ EXPECT_TRUE(
+ recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
}
TEST(AudioReceiveStreamTest, GetStats) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
- helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
+ helper.congestion_controller(), helper.config(), helper.audio_state());
helper.SetupMockForGetStats();
AudioReceiveStream::Stats stats = recv_stream.GetStats();
EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);