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https://github.com/mollyim/webrtc.git
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433eafe1f5
Bug: None Change-Id: Iadc531710dca0ba34a00ac363bfe0784355bb6f9 Reviewed-on: https://webrtc-review.googlesource.com/c/103501 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24995}
107 lines
3.6 KiB
C++
107 lines
3.6 KiB
C++
/*
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* Copyright (c) 2015 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/rtc_event_log_source.h"
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#include <string.h>
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#include <iostream>
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#include <limits>
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
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RtcEventLogSource* source = new RtcEventLogSource();
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RTC_CHECK(source->OpenFile(file_name));
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return source;
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}
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RtcEventLogSource::~RtcEventLogSource() {}
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bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) {
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RTC_CHECK(parser_.get());
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return parser_->RegisterRtpHeaderExtension(type, id);
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}
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std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
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for (; rtp_packet_index_ < parsed_stream_.GetNumberOfEvents();
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rtp_packet_index_++) {
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if (parsed_stream_.GetEventType(rtp_packet_index_) ==
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ParsedRtcEventLogNew::EventType::RTP_EVENT) {
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PacketDirection direction;
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size_t header_length;
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size_t packet_length;
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uint64_t timestamp_us = parsed_stream_.GetTimestamp(rtp_packet_index_);
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parsed_stream_.GetRtpHeader(rtp_packet_index_, &direction, nullptr,
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&header_length, &packet_length, nullptr);
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if (direction != kIncomingPacket) {
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continue;
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}
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uint8_t* packet_header = new uint8_t[header_length];
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parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, packet_header,
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nullptr, nullptr, nullptr);
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std::unique_ptr<Packet> packet(
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new Packet(packet_header, header_length, packet_length,
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static_cast<double>(timestamp_us) / 1000, *parser_.get()));
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if (!packet->valid_header()) {
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std::cout << "Warning: Packet with index " << rtp_packet_index_
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<< " has an invalid header and will be ignored." << std::endl;
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continue;
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}
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if (parsed_stream_.GetMediaType(packet->header().ssrc, direction) !=
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ParsedRtcEventLogNew::MediaType::AUDIO) {
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continue;
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}
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// Check if the packet should not be filtered out.
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if (!filter_.test(packet->header().payloadType) &&
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!(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
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++rtp_packet_index_;
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return packet;
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}
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}
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}
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return nullptr;
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}
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int64_t RtcEventLogSource::NextAudioOutputEventMs() {
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while (audio_output_index_ < parsed_stream_.GetNumberOfEvents()) {
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if (parsed_stream_.GetEventType(audio_output_index_) ==
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ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT) {
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LoggedAudioPlayoutEvent playout_event =
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parsed_stream_.GetAudioPlayout(audio_output_index_);
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if (!(use_ssrc_filter_ && playout_event.ssrc != ssrc_)) {
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audio_output_index_++;
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return playout_event.timestamp_us / 1000;
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}
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}
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audio_output_index_++;
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}
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return std::numeric_limits<int64_t>::max();
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}
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RtcEventLogSource::RtcEventLogSource()
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: PacketSource(), parser_(RtpHeaderParser::Create()) {}
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bool RtcEventLogSource::OpenFile(const std::string& file_name) {
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return parsed_stream_.ParseFile(file_name);
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}
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} // namespace test
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} // namespace webrtc
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