webrtc/modules/audio_coding/neteq/tools/neteq_input.cc
Jonas Olsson b2b2031457 Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00

93 lines
2.5 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