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Revert "Decrease complexity of RtpPacketHistory::GetBestFittingPacket."

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This reverts commit 54caa4b68a0acb81c7f6ef60ffec45b473a7e1a2.

Reason for revert: Crashes on some perf tests.
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fclient.webrtc.perf%2FLinux_Trusty%2F7170%2F%2B%2Frecipes%2Fsteps%2Fwebrtc_perf_tests%2F0%2Fstdout



Original change's description:
> Decrease complexity of RtpPacketHistory::GetBestFittingPacket.
> Use a map of packet sizes in RtpPacketHistory instead of looping through the whole history for every call.
> 
> Bug: webrtc:9731
> Change-Id: I44a4f6221e261a6cb3d5039edfa7556a102ee6f1
> Reviewed-on: https://webrtc-review.googlesource.com/98882
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24662}

TBR=danilchap@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: Id183cd31a67117e9614d163e4388131fd88de07d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9731
Reviewed-on: https://webrtc-review.googlesource.com/99440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24665}
This commit is contained in:
Per Kjellander
2018-09-11 05:34:57 +00:00
committed by Commit Bot
parent 3a66edf3c3
commit 49b2c3c4c4
3 changed files with 14 additions and 105 deletions
Download Patch File Download Diff File Expand all files Collapse all files

39
modules/rtp_rtcp/source/rtp_packet_history.cc
View File

@ -27,7 +27,8 @@ constexpr size_t kMinPacketRequestBytes = 50;
// Utility function to get the absolute difference in size between the provided
// target size and the size of packet.
size_t SizeDiff(size_t packet_size, size_t size) {
size_t SizeDiff(const std::unique_ptr<RtpPacketToSend>& packet, size_t size) {
size_t packet_size = packet->size();
if (packet_size > size) {
return packet_size - size;
}
@ -109,8 +110,6 @@ void RtpPacketHistory::PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
if (!start_seqno_) {
start_seqno_ = rtp_seq_no;
}
// Store the sequence number of the last send packet with this size.
packet_size_[stored_packet.packet->size()] = rtp_seq_no;
}
std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacketAndSetSendTime(
@ -187,34 +186,28 @@ std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetBestFittingPacket(
size_t packet_length) const {
// TODO(sprang): Make this smarter, taking retransmit count etc into account.
rtc::CritScope cs(&lock_);
if (packet_length < kMinPacketRequestBytes || packet_size_.empty()) {
if (packet_length < kMinPacketRequestBytes || packet_history_.empty()) {
return nullptr;
}
auto size_iter_upper = packet_size_.upper_bound(packet_length);
auto size_iter_lower = size_iter_upper;
if (size_iter_upper == packet_size_.end()) {
--size_iter_upper;
size_t min_diff = std::numeric_limits<size_t>::max();
RtpPacketToSend* best_packet = nullptr;
for (auto& it : packet_history_) {
size_t diff = SizeDiff(it.second.packet, packet_length);
if (!min_diff || diff < min_diff) {
min_diff = diff;
best_packet = it.second.packet.get();
if (diff == 0) {
break;
}
}
if (size_iter_lower != packet_size_.begin()) {
--size_iter_lower;
}
const size_t upper_bound_diff =
SizeDiff(size_iter_upper->first, packet_length);
const size_t lower_bound_diff =
SizeDiff(size_iter_lower->first, packet_length);
const uint16_t seq_no = upper_bound_diff < lower_bound_diff
? size_iter_upper->second
: size_iter_lower->second;
RtpPacketToSend* best_packet =
packet_history_.find(seq_no)->second.packet.get();
return absl::make_unique<RtpPacketToSend>(*best_packet);
}
void RtpPacketHistory::Reset() {
packet_history_.clear();
packet_size_.clear();
start_seqno_.reset();
}
@ -279,12 +272,6 @@ std::unique_ptr<RtpPacketToSend> RtpPacketHistory::RemovePacket(
start_seqno_.reset();
}
auto size_iterator = packet_size_.find(rtp_packet->size());
RTC_CHECK(size_iterator != packet_size_.end());
if (size_iterator->second == rtp_packet->SequenceNumber()) {
packet_size_.erase(size_iterator);
}
return rtp_packet;
}

1
modules/rtp_rtcp/source/rtp_packet_history.h
View File

@ -136,7 +136,6 @@ class RtpPacketHistory {
// Map from rtp sequence numbers to stored packet.
std::map<uint16_t, StoredPacket> packet_history_ RTC_GUARDED_BY(lock_);
std::map<size_t, uint16_t> packet_size_ RTC_GUARDED_BY(lock_);
// The earliest packet in the history. This might not be the lowest sequence
// number, in case there is a wraparound.

77
modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
View File

@ -16,7 +16,6 @@
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "system_wrappers/include/clock.h"
#include "test/gmock.h"
#include "test/gtest.h"
namespace webrtc {
@ -489,80 +488,4 @@ TEST_F(RtpPacketHistoryTest, GetBestFittingPacket) {
EXPECT_EQ(target_packet_size,
hist_.GetBestFittingPacket(target_packet_size)->size());
}
TEST_F(RtpPacketHistoryTest,
GetBestFittingPacketReturnsNextPacketWhenBestPacketHasBeenCulled) {
hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
packet->SetPayloadSize(50);
const size_t target_packet_size = packet->size();
hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
fake_clock_.TimeInMilliseconds());
packet = hist_.GetBestFittingPacket(target_packet_size + 2);
ASSERT_THAT(packet.get(), ::testing::NotNull());
// Send the packet and advance time past where packet expires.
ASSERT_THAT(hist_.GetPacketAndSetSendTime(kStartSeqNum, false),
::testing::NotNull());
fake_clock_.AdvanceTimeMilliseconds(
RtpPacketHistory::kPacketCullingDelayFactor *
RtpPacketHistory::kMinPacketDurationMs);
packet = CreateRtpPacket(kStartSeqNum + 1);
packet->SetPayloadSize(100);
hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
fake_clock_.TimeInMilliseconds());
ASSERT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
auto best_packet = hist_.GetBestFittingPacket(target_packet_size + 2);
ASSERT_THAT(best_packet.get(), ::testing::NotNull());
EXPECT_EQ(best_packet->SequenceNumber(), kStartSeqNum + 1);
}
TEST_F(RtpPacketHistoryTest, GetBestFittingPacketReturnLastPacketWhenSameSize) {
const size_t kTargetSize = 500;
hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
// Add two packets of same size.
std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
packet->SetPayloadSize(kTargetSize);
hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
fake_clock_.TimeInMilliseconds());
packet = CreateRtpPacket(kStartSeqNum + 1);
packet->SetPayloadSize(kTargetSize);
hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
fake_clock_.TimeInMilliseconds());
auto best_packet = hist_.GetBestFittingPacket(123);
ASSERT_THAT(best_packet.get(), ::testing::NotNull());
EXPECT_EQ(best_packet->SequenceNumber(), kStartSeqNum + 1);
}
TEST_F(RtpPacketHistoryTest,
GetBestFittingPacketReturnsPacketWithSmallestDiff) {
const size_t kTargetSize = 500;
hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
// Add two packets of very different size.
std::unique_ptr<RtpPacketToSend> small_packet = CreateRtpPacket(kStartSeqNum);
small_packet->SetPayloadSize(kTargetSize);
hist_.PutRtpPacket(std::move(small_packet), kAllowRetransmission,
fake_clock_.TimeInMilliseconds());
auto large_packet = CreateRtpPacket(kStartSeqNum + 1);
large_packet->SetPayloadSize(kTargetSize * 2);
hist_.PutRtpPacket(std::move(large_packet), kAllowRetransmission,
fake_clock_.TimeInMilliseconds());
ASSERT_THAT(hist_.GetBestFittingPacket(kTargetSize), ::testing::NotNull());
EXPECT_EQ(hist_.GetBestFittingPacket(kTargetSize)->SequenceNumber(),
kStartSeqNum);
ASSERT_THAT(hist_.GetBestFittingPacket(kTargetSize * 2),
::testing::NotNull());
EXPECT_EQ(hist_.GetBestFittingPacket(kTargetSize * 2)->SequenceNumber(),
kStartSeqNum + 1);
}
} // namespace webrtc