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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}
103 lines
3.1 KiB
C++
103 lines
3.1 KiB
C++
/*
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* Copyright (c) 2014 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/rtp_file_source.h"
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#include <assert.h>
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#include <string.h>
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#ifdef WIN32
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#include <winsock2.h>
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#else
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#include <netinet/in.h>
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#endif
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#include <memory>
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#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/test/rtp_file_reader.h"
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namespace webrtc {
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namespace test {
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RtpFileSource* RtpFileSource::Create(const std::string& file_name) {
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RtpFileSource* source = new RtpFileSource();
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RTC_CHECK(source->OpenFile(file_name));
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return source;
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}
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bool RtpFileSource::ValidRtpDump(const std::string& file_name) {
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std::unique_ptr<RtpFileReader> temp_file(
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RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
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return !!temp_file;
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}
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bool RtpFileSource::ValidPcap(const std::string& file_name) {
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std::unique_ptr<RtpFileReader> temp_file(
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RtpFileReader::Create(RtpFileReader::kPcap, file_name));
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return !!temp_file;
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}
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RtpFileSource::~RtpFileSource() {
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}
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bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) {
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assert(parser_.get());
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return parser_->RegisterRtpHeaderExtension(type, id);
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}
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std::unique_ptr<Packet> RtpFileSource::NextPacket() {
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while (true) {
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RtpPacket temp_packet;
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if (!rtp_reader_->NextPacket(&temp_packet)) {
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return NULL;
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}
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if (temp_packet.original_length == 0) {
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// May be an RTCP packet.
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// Read the next one.
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continue;
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}
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std::unique_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
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memcpy(packet_memory.get(), temp_packet.data, temp_packet.length);
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std::unique_ptr<Packet> packet(new Packet(
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packet_memory.release(), temp_packet.length,
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temp_packet.original_length, temp_packet.time_ms, *parser_.get()));
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if (!packet->valid_header()) {
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continue;
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}
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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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// This payload type should be filtered out. Continue to the next packet.
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continue;
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}
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return packet;
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}
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}
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RtpFileSource::RtpFileSource()
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: PacketSource(),
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parser_(RtpHeaderParser::Create()) {}
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bool RtpFileSource::OpenFile(const std::string& file_name) {
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rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
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if (rtp_reader_)
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return true;
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rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
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if (!rtp_reader_) {
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FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
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"that .pcapng is not supported.";
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}
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return true;
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}
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} // namespace test
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} // namespace webrtc
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