Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690
Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
Review URL: https://codereview.webrtc.org/1390753002 .
Cr-Commit-Position: refs/heads/master@{#10369}
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index c6f4b6b..d5061db 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -14,6 +14,8 @@
"audio_receive_stream.h",
"audio_send_stream.cc",
"audio_send_stream.h",
+ "conversion.h",
+ "scoped_voe_interface.h",
]
configs += [ "..:common_config" ]
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index c725e37..0fd96d0 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -12,10 +12,17 @@
#include <string>
+#include "webrtc/audio/conversion.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_neteq_stats.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include "webrtc/voice_engine/include/voe_video_sync.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
namespace webrtc {
std::string AudioReceiveStream::Config::Rtp::ToString() const {
@@ -24,8 +31,9 @@
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
- if (i != extensions.size() - 1)
+ if (i != extensions.size() - 1) {
ss << ", ";
+ }
}
ss << ']';
ss << '}';
@@ -36,8 +44,9 @@
std::stringstream ss;
ss << "{rtp: " << rtp.ToString();
ss << ", voe_channel_id: " << voe_channel_id;
- if (!sync_group.empty())
+ if (!sync_group.empty()) {
ss << ", sync_group: " << sync_group;
+ }
ss << '}';
return ss.str();
}
@@ -45,13 +54,18 @@
namespace internal {
AudioReceiveStream::AudioReceiveStream(
RemoteBitrateEstimator* remote_bitrate_estimator,
- const webrtc::AudioReceiveStream::Config& config)
+ const webrtc::AudioReceiveStream::Config& config,
+ VoiceEngine* voice_engine)
: remote_bitrate_estimator_(remote_bitrate_estimator),
config_(config),
+ voice_engine_(voice_engine),
+ voe_base_(voice_engine),
rtp_header_parser_(RtpHeaderParser::Create()) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
RTC_DCHECK(config.voe_channel_id != -1);
RTC_DCHECK(remote_bitrate_estimator_ != nullptr);
+ RTC_DCHECK(voice_engine_ != nullptr);
RTC_DCHECK(rtp_header_parser_ != nullptr);
for (const auto& ext : config.rtp.extensions) {
// One-byte-extension local identifiers are in the range 1-14 inclusive.
@@ -73,33 +87,117 @@
}
AudioReceiveStream::~AudioReceiveStream() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
}
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
- return webrtc::AudioReceiveStream::Stats();
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ webrtc::AudioReceiveStream::Stats stats;
+ stats.remote_ssrc = config_.rtp.remote_ssrc;
+ ScopedVoEInterface<VoECodec> codec(voice_engine_);
+ ScopedVoEInterface<VoENetEqStats> neteq(voice_engine_);
+ ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
+ ScopedVoEInterface<VoEVideoSync> sync(voice_engine_);
+ ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
+ unsigned int ssrc = 0;
+ webrtc::CallStatistics cs = {0};
+ webrtc::CodecInst ci = {0};
+ // Only collect stats if we have seen some traffic with the SSRC.
+ if (rtp->GetRemoteSSRC(config_.voe_channel_id, ssrc) == -1 ||
+ rtp->GetRTCPStatistics(config_.voe_channel_id, cs) == -1 ||
+ codec->GetRecCodec(config_.voe_channel_id, ci) == -1) {
+ return stats;
+ }
+
+ stats.bytes_rcvd = cs.bytesReceived;
+ stats.packets_rcvd = cs.packetsReceived;
+ stats.packets_lost = cs.cumulativeLost;
+ stats.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
+ if (ci.pltype != -1) {
+ stats.codec_name = ci.plname;
+ }
+
+ stats.ext_seqnum = cs.extendedMax;
+ if (ci.plfreq / 1000 > 0) {
+ stats.jitter_ms = cs.jitterSamples / (ci.plfreq / 1000);
+ }
+ {
+ int jitter_buffer_delay_ms = 0;
+ int playout_buffer_delay_ms = 0;
+ sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms,
+ &playout_buffer_delay_ms);
+ stats.delay_estimate_ms =
+ jitter_buffer_delay_ms + playout_buffer_delay_ms;
+ }
+ {
+ unsigned int level = 0;
+ if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level)
+ != -1) {
+ stats.audio_level = static_cast<int32_t>(level);
+ }
+ }
+
+ webrtc::NetworkStatistics ns = {0};
+ if (neteq->GetNetworkStatistics(config_.voe_channel_id, ns) != -1) {
+ // Get jitter buffer and total delay (alg + jitter + playout) stats.
+ stats.jitter_buffer_ms = ns.currentBufferSize;
+ stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
+ stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
+ stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
+ stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
+ stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
+ stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
+ }
+
+ webrtc::AudioDecodingCallStats ds;
+ if (neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds) != -1) {
+ stats.decoding_calls_to_silence_generator =
+ ds.calls_to_silence_generator;
+ stats.decoding_calls_to_neteq = ds.calls_to_neteq;
+ stats.decoding_normal = ds.decoded_normal;
+ stats.decoding_plc = ds.decoded_plc;
+ stats.decoding_cng = ds.decoded_cng;
+ stats.decoding_plc_cng = ds.decoded_plc_cng;
+ }
+
+ stats.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
+
+ return stats;
}
const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
return config_;
}
void AudioReceiveStream::Start() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
void AudioReceiveStream::Stop() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
void AudioReceiveStream::SignalNetworkState(NetworkState state) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+ // TODO(solenberg): Tests call this function on a network thread, libjingle
+ // calls on the worker thread. We should move towards always using a network
+ // thread. Then this check can be enabled.
+ // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
return false;
}
bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
+ // TODO(solenberg): Tests call this function on a network thread, libjingle
+ // calls on the worker thread. We should move towards always using a network
+ // thread. Then this check can be enabled.
+ // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
RTPHeader header;
if (!rtp_header_parser_->Parse(packet, length, &header)) {
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 1e52724..5c77653 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -12,18 +12,23 @@
#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
#include "webrtc/audio_receive_stream.h"
+#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/voice_engine/include/voe_base.h"
namespace webrtc {
class RemoteBitrateEstimator;
+class VoiceEngine;
namespace internal {
class AudioReceiveStream : public webrtc::AudioReceiveStream {
public:
AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
- const webrtc::AudioReceiveStream::Config& config);
+ const webrtc::AudioReceiveStream::Config& config,
+ VoiceEngine* voice_engine);
~AudioReceiveStream() override;
// webrtc::ReceiveStream implementation.
@@ -41,8 +46,12 @@
const webrtc::AudioReceiveStream::Config& config() const;
private:
+ rtc::ThreadChecker thread_checker_;
RemoteBitrateEstimator* const remote_bitrate_estimator_;
const webrtc::AudioReceiveStream::Config config_;
+ VoiceEngine* voice_engine_;
+ // We hold one interface pointer to the VoE to make sure it is kept alive.
+ ScopedVoEInterface<VoEBase> voe_base_;
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
};
} // namespace internal
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index d6cce69..8809b35 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -11,10 +11,14 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h"
+#include "webrtc/audio/conversion.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/test/fake_voice_engine.h"
-namespace webrtc {
+namespace {
+
+using webrtc::ByteWriter;
const size_t kAbsoluteSendTimeLength = 4;
@@ -45,33 +49,93 @@
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;
+ int32_t rtp_header_length = webrtc::kRtpHeaderSize;
BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
abs_send_time);
rtp_header_length += kAbsoluteSendTimeLength;
return rtp_header_length;
}
+} // namespace
+
+namespace webrtc {
+namespace test {
TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
- MockRemoteBitrateEstimator rbe;
+ MockRemoteBitrateEstimator remote_bitrate_estimator;
+ FakeVoiceEngine voice_engine;
AudioReceiveStream::Config config;
config.combined_audio_video_bwe = true;
- config.voe_channel_id = 1;
+ config.voe_channel_id = voice_engine.kReceiveChannelId;
const int kAbsSendTimeId = 3;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- internal::AudioReceiveStream recv_stream(&rbe, config);
+ internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
+ &voice_engine);
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))
+ EXPECT_CALL(remote_bitrate_estimator,
+ 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));
}
+
+TEST(AudioReceiveStreamTest, GetStats) {
+ const uint32_t kSsrc1 = 667;
+
+ MockRemoteBitrateEstimator remote_bitrate_estimator;
+ FakeVoiceEngine voice_engine;
+ AudioReceiveStream::Config config;
+ config.rtp.remote_ssrc = kSsrc1;
+ config.voe_channel_id = voice_engine.kReceiveChannelId;
+ internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
+ &voice_engine);
+
+ AudioReceiveStream::Stats stats = recv_stream.GetStats();
+ const CallStatistics& call_stats = voice_engine.GetRecvCallStats();
+ const CodecInst& codec_inst = voice_engine.GetRecvRecCodecInst();
+ const NetworkStatistics& net_stats = voice_engine.GetRecvNetworkStats();
+ const AudioDecodingCallStats& decode_stats =
+ voice_engine.GetRecvAudioDecodingCallStats();
+ EXPECT_EQ(kSsrc1, stats.remote_ssrc);
+ EXPECT_EQ(static_cast<int64_t>(call_stats.bytesReceived), stats.bytes_rcvd);
+ EXPECT_EQ(static_cast<uint32_t>(call_stats.packetsReceived),
+ stats.packets_rcvd);
+ EXPECT_EQ(call_stats.cumulativeLost, stats.packets_lost);
+ EXPECT_EQ(static_cast<float>(call_stats.fractionLost) / 256,
+ stats.fraction_lost);
+ EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
+ EXPECT_EQ(call_stats.extendedMax, stats.ext_seqnum);
+ EXPECT_EQ(call_stats.jitterSamples / (codec_inst.plfreq / 1000),
+ stats.jitter_ms);
+ EXPECT_EQ(net_stats.currentBufferSize, stats.jitter_buffer_ms);
+ EXPECT_EQ(net_stats.preferredBufferSize, stats.jitter_buffer_preferred_ms);
+ EXPECT_EQ(static_cast<uint32_t>(voice_engine.kRecvJitterBufferDelay +
+ voice_engine.kRecvPlayoutBufferDelay), stats.delay_estimate_ms);
+ EXPECT_EQ(static_cast<int32_t>(voice_engine.kRecvSpeechOutputLevel),
+ stats.audio_level);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentExpandRate), stats.expand_rate);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentSpeechExpandRate),
+ stats.speech_expand_rate);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentSecondaryDecodedRate),
+ stats.secondary_decoded_rate);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentAccelerateRate), stats.accelerate_rate);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentPreemptiveRate),
+ stats.preemptive_expand_rate);
+ EXPECT_EQ(decode_stats.calls_to_silence_generator,
+ stats.decoding_calls_to_silence_generator);
+ EXPECT_EQ(decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq);
+ EXPECT_EQ(decode_stats.decoded_normal, stats.decoding_normal);
+ EXPECT_EQ(decode_stats.decoded_plc, stats.decoding_plc);
+ EXPECT_EQ(decode_stats.decoded_cng, stats.decoding_cng);
+ EXPECT_EQ(decode_stats.decoded_plc_cng, stats.decoding_plc_cng);
+ EXPECT_EQ(call_stats.capture_start_ntp_time_ms_,
+ stats.capture_start_ntp_time_ms);
+}
+} // namespace test
} // namespace webrtc
diff --git a/webrtc/audio/conversion.h b/webrtc/audio/conversion.h
new file mode 100644
index 0000000..c1cf9b6
--- /dev/null
+++ b/webrtc/audio/conversion.h
@@ -0,0 +1,22 @@
+/*
+ * 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.
+ */
+
+#ifndef WEBRTC_AUDIO_CONVERSION_H_
+#define WEBRTC_AUDIO_CONVERSION_H_
+
+namespace webrtc {
+
+// Convert fixed point number with 14 bit fractional part, to floating point.
+inline float Q14ToFloat(uint16_t v) {
+ return static_cast<float>(v) / (1 << 14);
+}
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_CONVERSION_H_
diff --git a/webrtc/audio/scoped_voe_interface.h b/webrtc/audio/scoped_voe_interface.h
new file mode 100644
index 0000000..1029337
--- /dev/null
+++ b/webrtc/audio/scoped_voe_interface.h
@@ -0,0 +1,45 @@
+/*
+ * 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.
+ */
+
+#ifndef WEBRTC_AUDIO_SCOPED_VOE_INTERFACE_H_
+#define WEBRTC_AUDIO_SCOPED_VOE_INTERFACE_H_
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+namespace internal {
+
+// Utility template for obtaining and holding a reference to a VoiceEngine
+// interface and making sure it is released when this object goes out of scope.
+template<class T> class ScopedVoEInterface {
+ public:
+ explicit ScopedVoEInterface(webrtc::VoiceEngine* e)
+ : ptr_(T::GetInterface(e)) {
+ RTC_DCHECK(ptr_);
+ }
+ ~ScopedVoEInterface() {
+ if (ptr_) {
+ ptr_->Release();
+ }
+ }
+ T* operator->() {
+ RTC_DCHECK(ptr_);
+ return ptr_;
+ }
+ private:
+ T* ptr_;
+};
+} // namespace internal
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_SCOPED_VOE_INTERFACE_H_
diff --git a/webrtc/audio/webrtc_audio.gypi b/webrtc/audio/webrtc_audio.gypi
index 40ccff6..b9d45db 100644
--- a/webrtc/audio/webrtc_audio.gypi
+++ b/webrtc/audio/webrtc_audio.gypi
@@ -18,6 +18,8 @@
'audio/audio_receive_stream.h',
'audio/audio_send_stream.cc',
'audio/audio_send_stream.h',
+ 'audio/conversion.h',
+ 'audio/scoped_voe_interface.h',
],
},
}