Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.
This CL effectively reverts r6257 "Rename neteq4 folder to neteq".
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
new file mode 100644
index 0000000..8278635
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -0,0 +1,140 @@
+/*
+ * Copyright (c) 2014 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/rtp_file_source.h"
+
+#include <assert.h>
+#include <string.h>
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <netinet/in.h>
+#endif
+
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+
+namespace webrtc {
+namespace test {
+
+RtpFileSource* RtpFileSource::Create(const std::string& file_name) {
+ RtpFileSource* source = new RtpFileSource;
+ assert(source);
+ if (!source->OpenFile(file_name) || !source->SkipFileHeader()) {
+ assert(false);
+ delete source;
+ return NULL;
+ }
+ return source;
+}
+
+RtpFileSource::~RtpFileSource() {
+ if (in_file_)
+ fclose(in_file_);
+}
+
+bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
+ uint8_t id) {
+ assert(parser_.get());
+ return parser_->RegisterRtpHeaderExtension(type, id);
+}
+
+Packet* RtpFileSource::NextPacket() {
+ uint16_t length;
+ if (fread(&length, sizeof(uint16_t), 1, in_file_) == 0) {
+ assert(false);
+ return NULL;
+ }
+ length = ntohs(length);
+
+ uint16_t plen;
+ if (fread(&plen, sizeof(uint16_t), 1, in_file_) == 0) {
+ assert(false);
+ return NULL;
+ }
+ plen = ntohs(plen);
+
+ uint32_t offset;
+ if (fread(&offset, sizeof(uint32_t), 1, in_file_) == 0) {
+ assert(false);
+ return NULL;
+ }
+
+ // Use length here because a plen of 0 specifies RTCP.
+ size_t packet_size_bytes = length - kPacketHeaderSize;
+ if (packet_size_bytes <= 0) {
+ // May be an RTCP packet.
+ return NULL;
+ }
+ uint8_t* packet_memory = new uint8_t[packet_size_bytes];
+ if (fread(packet_memory, 1, packet_size_bytes, in_file_) !=
+ packet_size_bytes) {
+ assert(false);
+ delete[] packet_memory;
+ return NULL;
+ }
+ Packet* packet = new Packet(
+ packet_memory, packet_size_bytes, plen, ntohl(offset), *parser_.get());
+ if (!packet->valid_header()) {
+ assert(false);
+ delete packet;
+ return NULL;
+ }
+ return packet;
+}
+
+bool RtpFileSource::EndOfFile() const {
+ assert(in_file_);
+ return ftell(in_file_) >= file_end_;
+}
+
+RtpFileSource::RtpFileSource()
+ : PacketSource(),
+ in_file_(NULL),
+ file_end_(-1),
+ parser_(RtpHeaderParser::Create()) {}
+
+bool RtpFileSource::OpenFile(const std::string& file_name) {
+ in_file_ = fopen(file_name.c_str(), "rb");
+ assert(in_file_);
+ if (in_file_ == NULL) {
+ return false;
+ }
+
+ // Find out how long the file is.
+ fseek(in_file_, 0, SEEK_END);
+ file_end_ = ftell(in_file_);
+ rewind(in_file_);
+ return true;
+}
+
+bool RtpFileSource::SkipFileHeader() {
+ char firstline[kFirstLineLength];
+ assert(in_file_);
+ if (fgets(firstline, kFirstLineLength, in_file_) == NULL) {
+ assert(false);
+ return false;
+ }
+ // Check that the first line is ok.
+ if ((strncmp(firstline, "#!rtpplay1.0", 12) != 0) &&
+ (strncmp(firstline, "#!RTPencode1.0", 14) != 0)) {
+ assert(false);
+ return false;
+ }
+ // Skip the file header.
+ if (fseek(in_file_, kRtpFileHeaderSize, SEEK_CUR) != 0) {
+ assert(false);
+ return false;
+ }
+ return true;
+}
+
+} // namespace test
+} // namespace webrtc
