Delete unused NetEq Rtcp stats.
Bug: webrtc:7135
Change-Id: Ib3ca9e02b051b8b41c2eac4e43a4f1f37999bf75
Reviewed-on: https://webrtc-review.googlesource.com/c/111640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25743}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 5163d16..3422665 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1025,8 +1025,6 @@
"neteq/random_vector.h",
"neteq/red_payload_splitter.cc",
"neteq/red_payload_splitter.h",
- "neteq/rtcp.cc",
- "neteq/rtcp.h",
"neteq/statistics_calculator.cc",
"neteq/statistics_calculator.h",
"neteq/sync_buffer.cc",
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index 530975f..3b3d969 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -231,13 +231,6 @@
// statistics are never reset.
virtual NetEqOperationsAndState GetOperationsAndState() const = 0;
- // Writes the current RTCP statistics to |stats|. The statistics are reset
- // and a new report period is started with the call.
- virtual void GetRtcpStatistics(RtcpStatistics* stats) = 0;
-
- // Same as RtcpStatistics(), but does not reset anything.
- virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats) = 0;
-
// Enables post-decode VAD. When enabled, GetAudio() will return
// kOutputVADPassive when the signal contains no speech.
virtual void EnableVad() = 0;
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 6a2cbae..68af21a 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -387,20 +387,6 @@
return result;
}
-void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) {
- rtc::CritScope lock(&crit_sect_);
- if (stats) {
- rtcp_.GetStatistics(false, stats);
- }
-}
-
-void NetEqImpl::GetRtcpStatisticsNoReset(RtcpStatistics* stats) {
- rtc::CritScope lock(&crit_sect_);
- if (stats) {
- rtcp_.GetStatistics(true, stats);
- }
-}
-
void NetEqImpl::EnableVad() {
rtc::CritScope lock(&crit_sect_);
assert(vad_.get());
@@ -576,8 +562,6 @@
// Note: |first_packet_| will be cleared further down in this method, once
// the packet has been successfully inserted into the packet buffer.
- rtcp_.Init(rtp_header.sequenceNumber);
-
// Flush the packet buffer and DTMF buffer.
packet_buffer_->Flush();
dtmf_buffer_->Flush();
@@ -592,9 +576,6 @@
timestamp_ = main_timestamp;
}
- // Update RTCP statistics, only for regular packets.
- rtcp_.Update(rtp_header, receive_timestamp);
-
if (nack_enabled_) {
RTC_DCHECK(nack_);
if (update_sample_rate_and_channels) {
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 36990fb..6fcaf9c 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -22,7 +22,6 @@
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/packet.h"
#include "modules/audio_coding/neteq/random_vector.h"
-#include "modules/audio_coding/neteq/rtcp.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/constructormagic.h"
@@ -170,17 +169,10 @@
// after the call.
int NetworkStatistics(NetEqNetworkStatistics* stats) override;
- // Writes the current RTCP statistics to |stats|. The statistics are reset
- // and a new report period is started with the call.
- void GetRtcpStatistics(RtcpStatistics* stats) override;
-
NetEqLifetimeStatistics GetLifetimeStatistics() const override;
NetEqOperationsAndState GetOperationsAndState() const override;
- // Same as RtcpStatistics(), but does not reset anything.
- void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
-
// Enables post-decode VAD. When enabled, GetAudio() will return
// kOutputVADPassive when the signal contains no speech.
void EnableVad() override;
@@ -395,7 +387,6 @@
RTC_GUARDED_BY(crit_sect_);
RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
- Rtcp rtcp_ RTC_GUARDED_BY(crit_sect_);
StatisticsCalculator stats_ RTC_GUARDED_BY(crit_sect_);
int fs_hz_ RTC_GUARDED_BY(crit_sect_);
int fs_mult_ RTC_GUARDED_BY(crit_sect_);
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 2130a24..2aced6a 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -265,7 +265,6 @@
void DecodeAndCompare(const std::string& rtp_file,
const std::string& output_checksum,
const std::string& network_stats_checksum,
- const std::string& rtcp_stats_checksum,
bool gen_ref);
static void PopulateRtpInfo(int frame_index,
@@ -373,7 +372,6 @@
const std::string& rtp_file,
const std::string& output_checksum,
const std::string& network_stats_checksum,
- const std::string& rtcp_stats_checksum,
bool gen_ref) {
OpenInputFile(rtp_file);
@@ -385,10 +383,6 @@
gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : "";
ResultSink network_stats(stat_out_file);
- std::string rtcp_out_file =
- gen_ref ? webrtc::test::OutputPath() + "neteq_rtcp_stats.dat" : "";
- ResultSink rtcp_stats(rtcp_out_file);
-
packet_ = rtp_source_->NextPacket();
int i = 0;
uint64_t last_concealed_samples = 0;
@@ -425,11 +419,6 @@
EXPECT_NEAR(
(delta_concealed_samples << 14) / delta_total_samples_received,
current_network_stats.expand_rate, (2 << 14) / 100.0);
-
- // Process RTCPstat.
- RtcpStatistics current_rtcp_stats;
- neteq_->GetRtcpStatistics(¤t_rtcp_stats);
- ASSERT_NO_FATAL_FAILURE(rtcp_stats.AddResult(current_rtcp_stats));
}
}
@@ -437,8 +426,6 @@
output.VerifyChecksum(output_checksum);
SCOPED_TRACE("Check network stats.");
network_stats.VerifyChecksum(network_stats_checksum);
- SCOPED_TRACE("Check rtcp stats.");
- rtcp_stats.VerifyChecksum(rtcp_stats_checksum);
}
void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
@@ -488,14 +475,8 @@
"4b2370f5c794741d2a46be5c7935c66ef3fb53e9",
"4b2370f5c794741d2a46be5c7935c66ef3fb53e9");
- const std::string rtcp_stats_checksum =
- PlatformChecksum("b8880bf9fed2487efbddcb8d94b9937a29ae521d",
- "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4", "not used",
- "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
- "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
-
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
- rtcp_stats_checksum, FLAG_gen_ref);
+ FLAG_gen_ref);
}
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
@@ -523,15 +504,8 @@
"adb3272498e436d1c019cbfd71610e9510c54497",
"adb3272498e436d1c019cbfd71610e9510c54497");
- const std::string rtcp_stats_checksum =
- PlatformChecksum("e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
- "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
- "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
- "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
- "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
-
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
- rtcp_stats_checksum, FLAG_gen_ref);
+ FLAG_gen_ref);
}
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
@@ -554,11 +528,8 @@
const std::string network_stats_checksum =
"bab58dc587d956f326056d7340c96eb9d2d3cc21";
- const std::string rtcp_stats_checksum =
- "ac27a7f305efb58b39bf123dccee25dee5758e63";
-
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
- rtcp_stats_checksum, FLAG_gen_ref);
+ FLAG_gen_ref);
}
// Use fax mode to avoid time-scaling. This is to simplify the testing of
diff --git a/modules/audio_coding/neteq/rtcp.cc b/modules/audio_coding/neteq/rtcp.cc
deleted file mode 100644
index 6519337..0000000
--- a/modules/audio_coding/neteq/rtcp.cc
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyright (c) 2011 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 "modules/audio_coding/neteq/rtcp.h"
-
-#include <algorithm>
-#include <cstdlib>
-
-#include "api/rtp_headers.h"
-#include "common_types.h"
-#include "rtc_base/checks.h"
-
-namespace webrtc {
-
-void Rtcp::Init(uint16_t start_sequence_number) {
- cycles_ = 0;
- max_seq_no_ = start_sequence_number;
- base_seq_no_ = start_sequence_number;
- received_packets_ = 0;
- received_packets_prior_ = 0;
- expected_prior_ = 0;
- jitter_ = 0;
- transit_ = 0;
-}
-
-void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) {
- // Update number of received packets, and largest packet number received.
- received_packets_++;
- int16_t sn_diff = rtp_header.sequenceNumber - max_seq_no_;
- if (sn_diff >= 0) {
- if (rtp_header.sequenceNumber < max_seq_no_) {
- // Wrap-around detected.
- cycles_++;
- }
- max_seq_no_ = rtp_header.sequenceNumber;
- }
-
- // Calculate jitter according to RFC 3550, and update previous timestamps.
- // Note that the value in |jitter_| is in Q4.
- if (received_packets_ > 1) {
- int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_);
- int64_t jitter_diff = (std::abs(int64_t{ts_diff}) << 4) - jitter_;
- // Calculate 15 * jitter_ / 16 + jitter_diff / 16 (with proper rounding).
- jitter_ = jitter_ + ((jitter_diff + 8) >> 4);
- RTC_DCHECK_GE(jitter_, 0);
- }
- transit_ = rtp_header.timestamp - receive_timestamp;
-}
-
-void Rtcp::GetStatistics(bool no_reset, RtcpStatistics* stats) {
- // Extended highest sequence number received.
- stats->extended_highest_sequence_number =
- (static_cast<int>(cycles_) << 16) + max_seq_no_;
-
- // Calculate expected number of packets and compare it with the number of
- // packets that were actually received. The cumulative number of lost packets
- // can be extracted.
- uint32_t expected_packets =
- stats->extended_highest_sequence_number - base_seq_no_ + 1;
- if (received_packets_ == 0) {
- // No packets received, assume none lost.
- stats->packets_lost = 0;
- } else if (expected_packets > received_packets_) {
- stats->packets_lost = expected_packets - received_packets_;
- if (stats->packets_lost > 0xFFFFFF) {
- stats->packets_lost = 0xFFFFFF;
- }
- } else {
- stats->packets_lost = 0;
- }
-
- // Fraction lost since last report.
- uint32_t expected_since_last = expected_packets - expected_prior_;
- uint32_t received_since_last = received_packets_ - received_packets_prior_;
- if (!no_reset) {
- expected_prior_ = expected_packets;
- received_packets_prior_ = received_packets_;
- }
- int32_t lost = expected_since_last - received_since_last;
- if (expected_since_last == 0 || lost <= 0 || received_packets_ == 0) {
- stats->fraction_lost = 0;
- } else {
- stats->fraction_lost = std::min(0xFFU, (lost << 8) / expected_since_last);
- }
-
- stats->jitter = jitter_ >> 4; // Scaling from Q4.
-}
-
-} // namespace webrtc
diff --git a/modules/audio_coding/neteq/rtcp.h b/modules/audio_coding/neteq/rtcp.h
deleted file mode 100644
index 60c2673..0000000
--- a/modules/audio_coding/neteq/rtcp.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 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 MODULES_AUDIO_CODING_NETEQ_RTCP_H_
-#define MODULES_AUDIO_CODING_NETEQ_RTCP_H_
-
-#include <stdint.h>
-
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-struct RtcpStatistics;
-struct RTPHeader;
-
-class Rtcp {
- public:
- Rtcp() { Init(0); }
-
- ~Rtcp() {}
-
- // Resets the RTCP statistics, and sets the first received sequence number.
- void Init(uint16_t start_sequence_number);
-
- // Updates the RTCP statistics with a new received packet.
- void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp);
-
- // Returns the current RTCP statistics. If |no_reset| is true, the statistics
- // are not reset, otherwise they are.
- void GetStatistics(bool no_reset, RtcpStatistics* stats);
-
- private:
- uint16_t cycles_; // The number of wrap-arounds for the sequence number.
- uint16_t max_seq_no_; // The maximum sequence number received. Starts over
- // from 0 after wrap-around.
- uint16_t base_seq_no_; // The sequence number of the first received packet.
- uint32_t received_packets_; // The number of packets that have been received.
- uint32_t received_packets_prior_; // Number of packets received when last
- // report was generated.
- uint32_t expected_prior_; // Expected number of packets, at the time of the
- // last report.
- int64_t jitter_; // Current jitter value in Q4.
- int32_t transit_; // Clock difference for previous packet.
-
- RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp);
-};
-
-} // namespace webrtc
-#endif // MODULES_AUDIO_CODING_NETEQ_RTCP_H_