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1d87b0e40f
We're moving to an RtcEventLog interface that accepts std::unique_ptr<EventLog> and stores the event for encoding when encoding becomes necessary, rather than before. This will be useful while we maintain the legacy (current) encoding alongside the new encoding on which we're working. This CL introduces RtcEventLogEncoderLegacy, which takes provides the encoding currently done by RtcEventLogImpl. After this, we can modify RtcEventLogImpl to use a dynamically chosen encoding, allowing us to easily choose between the current encoding and the new one on which we're working. BUG=webrtc:8111 TBR=stefan@webrtc.org Change-Id: I3dde7e222a40a117549a094a59b04219467f490a Reviewed-on: https://webrtc-review.googlesource.com/1364 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20116}
109 lines
3.7 KiB
C++
109 lines
3.7 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 <assert.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 "call/call.h"
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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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ParsedRtcEventLog::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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webrtc::ParsedRtcEventLog::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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ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT) {
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uint64_t timestamp_us = parsed_stream_.GetTimestamp(audio_output_index_);
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// We call GetAudioPlayout only to check that the protobuf event is
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// well-formed.
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parsed_stream_.GetAudioPlayout(audio_output_index_, nullptr);
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audio_output_index_++;
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return timestamp_us / 1000;
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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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