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/*
* 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() {
while (!EndOfFile()) {
uint16_t length;
if (fread(&length, sizeof(length), 1, in_file_) == 0) {
assert(false);
return NULL;
}
length = ntohs(length);
uint16_t plen;
if (fread(&plen, sizeof(plen), 1, in_file_) == 0) {
assert(false);
return NULL;
}
plen = ntohs(plen);
uint32_t offset;
if (fread(&offset, sizeof(offset), 1, in_file_) == 0) {
assert(false);
return NULL;
}
offset = ntohl(offset);
// Use length here because a plen of 0 specifies RTCP.
assert(length >= kPacketHeaderSize);
size_t packet_size_bytes = length - kPacketHeaderSize;
if (packet_size_bytes == 0) {
// May be an RTCP packet.
// Read the next one.
continue;
}
scoped_ptr<uint8_t> packet_memory(new uint8_t[packet_size_bytes]);
if (fread(packet_memory.get(), 1, packet_size_bytes, in_file_) !=
packet_size_bytes) {
assert(false);
return NULL;
}
scoped_ptr<Packet> packet(new Packet(packet_memory.release(),
packet_size_bytes,
plen,
offset,
*parser_.get()));
if (!packet->valid_header()) {
assert(false);
return NULL;
}
return packet.release();
}
return NULL;
}
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
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