Implement NetEq's CurrentDelay function
This was not implemented before. It returns the current total delay (packet buffer and sync buffer) of NetEq. This is the same information that was already available in NetEqNetworkStatistics::current_buffer_size_ms, that can be obtained through NetEq::NetworkStatistics(). But, since the current delay is a key metric of NetEq, it is convenient to have it available in a simpler way.
R=kwiberg@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51149004
Cr-Commit-Position: refs/heads/master@{#9359}
diff --git a/webrtc/modules/audio_coding/neteq/interface/neteq.h b/webrtc/modules/audio_coding/neteq/interface/neteq.h
index 88bf208..93e8172 100644
--- a/webrtc/modules/audio_coding/neteq/interface/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/interface/neteq.h
@@ -210,8 +210,8 @@
// Not implemented.
virtual int TargetDelay() = 0;
- // Not implemented.
- virtual int CurrentDelay() = 0;
+ // Returns the current total delay (packet buffer and sync buffer) in ms.
+ virtual int CurrentDelayMs() const = 0;
// Sets the playout mode to |mode|.
// Deprecated. Set the mode in the Config struct passed to the constructor.
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 1351e66..636594e 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -15,6 +15,7 @@
#include <algorithm>
+#include "webrtc/base/checks.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/accelerate.h"
@@ -291,8 +292,19 @@
return kNotImplemented;
}
-int NetEqImpl::CurrentDelay() {
- return kNotImplemented;
+int NetEqImpl::CurrentDelayMs() const {
+ CriticalSectionScoped lock(crit_sect_.get());
+ if (fs_hz_ == 0)
+ return 0;
+ // Sum up the samples in the packet buffer with the future length of the sync
+ // buffer, and divide the sum by the sample rate.
+ const int delay_samples =
+ packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
+ decoder_frame_length_) +
+ static_cast<int>(sync_buffer_->FutureLength());
+ // The division below will truncate.
+ const int delay_ms = delay_samples / rtc::CheckedDivExact(fs_hz_, 1000);
+ return delay_ms;
}
// Deprecated.
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 55ba067..4c4660a 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -138,7 +138,7 @@
int TargetDelay() override;
- int CurrentDelay() override;
+ int CurrentDelayMs() const override;
// Sets the playout mode to |mode|.
// Deprecated.
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index e1a0f69..95f6489 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -170,6 +170,9 @@
CHECK_NETEQ_NETWORK_STATS(added_zero_samples);
#undef CHECK_NETEQ_NETWORK_STATS
+
+ // Compare with CurrentDelay, which should be identical.
+ EXPECT_EQ(stats.current_buffer_size_ms, neteq()->CurrentDelayMs());
}
void RunTest(int num_loops, NetEqNetworkStatsCheck expects) {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index b3d6f25..678e3a0 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -404,6 +404,8 @@
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
ASSERT_NO_FATAL_FAILURE(
network_stat_files.ProcessReference(network_stats));
+ // Compare with CurrentDelay, which should be identical.
+ EXPECT_EQ(network_stats.current_buffer_size_ms, neteq_->CurrentDelayMs());
// Process RTCPstat.
RtcpStatistics rtcp_stats;