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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}
93 lines
2.5 KiB
C++
93 lines
2.5 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/tools/neteq_input.h"
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#include "rtc_base/strings/string_builder.h"
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namespace webrtc {
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namespace test {
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NetEqInput::PacketData::PacketData() = default;
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NetEqInput::PacketData::~PacketData() = default;
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std::string NetEqInput::PacketData::ToString() const {
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rtc::StringBuilder ss;
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ss << "{"
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"time_ms: "
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<< static_cast<int64_t>(time_ms)
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<< ", "
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"header: {"
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"pt: "
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<< static_cast<int>(header.payloadType)
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<< ", "
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"sn: "
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<< header.sequenceNumber
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<< ", "
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"ts: "
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<< header.timestamp
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<< ", "
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"ssrc: "
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<< header.ssrc
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<< "}, "
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"payload bytes: "
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<< payload.size() << "}";
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return ss.Release();
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}
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TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
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int64_t duration_ms)
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: input_(std::move(input)),
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start_time_ms_(input_->NextEventTime()),
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duration_ms_(duration_ms) {}
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TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;
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absl::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
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return ended_ ? absl::nullopt : input_->NextPacketTime();
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}
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absl::optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
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return ended_ ? absl::nullopt : input_->NextOutputEventTime();
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}
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std::unique_ptr<NetEqInput::PacketData> TimeLimitedNetEqInput::PopPacket() {
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if (ended_) {
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return std::unique_ptr<PacketData>();
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}
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auto packet = input_->PopPacket();
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MaybeSetEnded();
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return packet;
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}
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void TimeLimitedNetEqInput::AdvanceOutputEvent() {
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if (!ended_) {
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input_->AdvanceOutputEvent();
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MaybeSetEnded();
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}
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}
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bool TimeLimitedNetEqInput::ended() const {
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return ended_ || input_->ended();
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}
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absl::optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
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return ended_ ? absl::nullopt : input_->NextHeader();
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}
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void TimeLimitedNetEqInput::MaybeSetEnded() {
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if (NextEventTime() && start_time_ms_ &&
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*NextEventTime() - *start_time_ms_ > duration_ms_) {
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ended_ = true;
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}
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}
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} // namespace test
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} // namespace webrtc
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