webrtc/modules/audio_coding/neteq/tools/neteq_input.cc
Jonas Olsson 84df1c724e Make fewer copies when using StringBuilder.
Replace calls to .str() which copies with .Release which moves in cases where that's safe.

This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"

Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
2018-09-24 09:39:19 +00:00

82 lines
2.4 KiB
C++

/*
* Copyright (c) 2017 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/tools/neteq_input.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
namespace test {
NetEqInput::PacketData::PacketData() = default;
NetEqInput::PacketData::~PacketData() = default;
std::string NetEqInput::PacketData::ToString() const {
rtc::StringBuilder ss;
ss << "{"
<< "time_ms: " << static_cast<int64_t>(time_ms) << ", "
<< "header: {"
<< "pt: " << static_cast<int>(header.payloadType) << ", "
<< "sn: " << header.sequenceNumber << ", "
<< "ts: " << header.timestamp << ", "
<< "ssrc: " << header.ssrc << "}, "
<< "payload bytes: " << payload.size() << "}";
return ss.Release();
}
TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
int64_t duration_ms)
: input_(std::move(input)),
start_time_ms_(input_->NextEventTime()),
duration_ms_(duration_ms) {}
TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;
absl::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
return ended_ ? absl::nullopt : input_->NextPacketTime();
}
absl::optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
return ended_ ? absl::nullopt : input_->NextOutputEventTime();
}
std::unique_ptr<NetEqInput::PacketData> TimeLimitedNetEqInput::PopPacket() {
if (ended_) {
return std::unique_ptr<PacketData>();
}
auto packet = input_->PopPacket();
MaybeSetEnded();
return packet;
}
void TimeLimitedNetEqInput::AdvanceOutputEvent() {
if (!ended_) {
input_->AdvanceOutputEvent();
MaybeSetEnded();
}
}
bool TimeLimitedNetEqInput::ended() const {
return ended_ || input_->ended();
}
absl::optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
return ended_ ? absl::nullopt : input_->NextHeader();
}
void TimeLimitedNetEqInput::MaybeSetEnded() {
if (NextEventTime() && start_time_ms_ &&
*NextEventTime() - *start_time_ms_ > duration_ms_) {
ended_ = true;
}
}
} // namespace test
} // namespace webrtc