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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
116 lines
3.7 KiB
C++
116 lines
3.7 KiB
C++
/*
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* Copyright (c) 2016 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 "webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
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#include <algorithm>
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#include <limits>
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#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
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#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
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#include "webrtc/rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {}
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rtc::Optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
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return packet_
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? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
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: rtc::Optional<int64_t>();
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}
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rtc::Optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
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return packet_ ? rtc::Optional<RTPHeader>(packet_->header())
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: rtc::Optional<RTPHeader>();
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}
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void NetEqPacketSourceInput::LoadNextPacket() {
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packet_ = source()->NextPacket();
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}
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std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() {
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if (!packet_) {
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return std::unique_ptr<PacketData>();
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}
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std::unique_ptr<PacketData> packet_data(new PacketData);
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packet_data->header = packet_->header();
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if (packet_->payload_length_bytes() == 0 &&
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packet_->virtual_payload_length_bytes() > 0) {
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// This is a header-only "dummy" packet. Set the payload to all zeros, with
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// length according to the virtual length.
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packet_data->payload.SetSize(packet_->virtual_payload_length_bytes());
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std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
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} else {
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packet_data->payload.SetData(packet_->payload(),
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packet_->payload_length_bytes());
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}
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packet_data->time_ms = packet_->time_ms();
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LoadNextPacket();
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return packet_data;
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}
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NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name,
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const RtpHeaderExtensionMap& hdr_ext_map)
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: source_(RtpFileSource::Create(file_name)) {
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for (const auto& ext_pair : hdr_ext_map) {
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source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
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}
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LoadNextPacket();
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}
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rtc::Optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
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return next_output_event_ms_;
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}
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void NetEqRtpDumpInput::AdvanceOutputEvent() {
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if (next_output_event_ms_) {
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*next_output_event_ms_ += kOutputPeriodMs;
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}
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if (!NextPacketTime()) {
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next_output_event_ms_ = rtc::Optional<int64_t>();
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}
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}
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PacketSource* NetEqRtpDumpInput::source() {
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return source_.get();
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}
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NetEqEventLogInput::NetEqEventLogInput(const std::string& file_name,
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const RtpHeaderExtensionMap& hdr_ext_map)
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: source_(RtcEventLogSource::Create(file_name)) {
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for (const auto& ext_pair : hdr_ext_map) {
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source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
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}
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LoadNextPacket();
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AdvanceOutputEvent();
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}
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rtc::Optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
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return rtc::Optional<int64_t>(next_output_event_ms_);
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}
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void NetEqEventLogInput::AdvanceOutputEvent() {
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next_output_event_ms_ =
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rtc::Optional<int64_t>(source_->NextAudioOutputEventMs());
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if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
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next_output_event_ms_ = rtc::Optional<int64_t>();
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}
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}
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PacketSource* NetEqEventLogInput::source() {
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return source_.get();
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}
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} // namespace test
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} // namespace webrtc
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