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Reland: Implement NetEq's CurrentDelay function

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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.

This is a re-landing of r9359,
https://webrtc-codereview.appspot.com/51149004, which was reverted in
r9360. The refactoring made in r9669 facilitated the relanding.

TBR=minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1313873003

Cr-Commit-Position: refs/heads/master@{#9801}
This commit is contained in:
henrik.lundin 2015-08-27 13:12:22 -07:00 committed by Commit bot
parent a567bf3292
commit 9c3efd0052
5 changed files with 23 additions and 5 deletions
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4
webrtc/modules/audio_coding/neteq/interface/neteq.h
View File

@ -215,8 +215,8 @@ class NetEq {
// 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.

17
webrtc/modules/audio_coding/neteq/neteq_impl.cc
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@ -15,6 +15,7 @@
#include <algorithm>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
@ -282,8 +283,20 @@ int NetEqImpl::TargetDelay() {
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 size_t delay_samples =
packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
decoder_frame_length_) +
sync_buffer_->FutureLength();
// The division below will truncate.
const int delay_ms =
static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
return delay_ms;
}
// Deprecated.

2
webrtc/modules/audio_coding/neteq/neteq_impl.h
View File

@ -136,7 +136,7 @@ class NetEqImpl : public webrtc::NetEq {
int TargetDelay() override;
int CurrentDelay() override;
int CurrentDelayMs() const override;
// Sets the playout mode to |mode|.
// Deprecated.

3
webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
View File

@ -170,6 +170,9 @@ struct NetEqNetworkStatsCheck {
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) {

2
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -406,6 +406,8 @@ void NetEqDecodingTest::DecodeAndCompare(const std::string& rtp_file,
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;