| /* |
| * Copyright (c) 2016 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 "webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h" |
| |
| #include <algorithm> |
| #include <limits> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {} |
| |
| rtc::Optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const { |
| return packet_ |
| ? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms())) |
| : rtc::Optional<int64_t>(); |
| } |
| |
| rtc::Optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const { |
| return packet_ ? rtc::Optional<RTPHeader>(packet_->header()) |
| : rtc::Optional<RTPHeader>(); |
| } |
| |
| void NetEqPacketSourceInput::LoadNextPacket() { |
| packet_ = source()->NextPacket(); |
| } |
| |
| std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() { |
| if (!packet_) { |
| return std::unique_ptr<PacketData>(); |
| } |
| std::unique_ptr<PacketData> packet_data(new PacketData); |
| packet_->ConvertHeader(&packet_data->header); |
| if (packet_->payload_length_bytes() == 0 && |
| packet_->virtual_payload_length_bytes() > 0) { |
| // This is a header-only "dummy" packet. Set the payload to all zeros, with |
| // length according to the virtual length. |
| packet_data->payload.SetSize(packet_->virtual_payload_length_bytes()); |
| std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0); |
| } else { |
| packet_data->payload.SetData(packet_->payload(), |
| packet_->payload_length_bytes()); |
| } |
| packet_data->time_ms = packet_->time_ms(); |
| |
| LoadNextPacket(); |
| |
| return packet_data; |
| } |
| |
| NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name, |
| const RtpHeaderExtensionMap& hdr_ext_map) |
| : source_(RtpFileSource::Create(file_name)) { |
| for (const auto& ext_pair : hdr_ext_map) { |
| source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first); |
| } |
| LoadNextPacket(); |
| } |
| |
| rtc::Optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const { |
| return next_output_event_ms_; |
| } |
| |
| void NetEqRtpDumpInput::AdvanceOutputEvent() { |
| if (next_output_event_ms_) { |
| *next_output_event_ms_ += kOutputPeriodMs; |
| } |
| if (!NextPacketTime()) { |
| next_output_event_ms_ = rtc::Optional<int64_t>(); |
| } |
| } |
| |
| PacketSource* NetEqRtpDumpInput::source() { |
| return source_.get(); |
| } |
| |
| NetEqEventLogInput::NetEqEventLogInput(const std::string& file_name, |
| const RtpHeaderExtensionMap& hdr_ext_map) |
| : source_(RtcEventLogSource::Create(file_name)) { |
| for (const auto& ext_pair : hdr_ext_map) { |
| source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first); |
| } |
| LoadNextPacket(); |
| AdvanceOutputEvent(); |
| } |
| |
| rtc::Optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const { |
| return rtc::Optional<int64_t>(next_output_event_ms_); |
| } |
| |
| void NetEqEventLogInput::AdvanceOutputEvent() { |
| next_output_event_ms_ = |
| rtc::Optional<int64_t>(source_->NextAudioOutputEventMs()); |
| if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) { |
| next_output_event_ms_ = rtc::Optional<int64_t>(); |
| } |
| } |
| |
| PacketSource* NetEqEventLogInput::source() { |
| return source_.get(); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |