webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
Mirko Bonadei a4fd641f51 Revert "Rename FATAL() into RTC_FATAL()."
This reverts commit 9653d26f8e.

Reason for revert: Breaks downstream project.

Original change's description:
> Rename FATAL() into RTC_FATAL().
>
> No-Try: True
> Bug: webrtc:8454
> Change-Id: I9130487a92463a2128cf1493e6c5117b2fab313a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193703
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32620}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I0ad01bcac60c87b30bd4575a9d631e7dd8f34992
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8454
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193863
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32627}
2020-11-18 07:03:54 +00:00

101 lines
3.2 KiB
C++

/*
* 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 "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include <assert.h>
#include <string.h>
#ifndef WIN32
#include <netinet/in.h>
#endif
#include <memory>
#include "modules/audio_coding/neteq/tools/packet.h"
#include "rtc_base/checks.h"
#include "test/rtp_file_reader.h"
namespace webrtc {
namespace test {
RtpFileSource* RtpFileSource::Create(const std::string& file_name,
absl::optional<uint32_t> ssrc_filter) {
RtpFileSource* source = new RtpFileSource(ssrc_filter);
RTC_CHECK(source->OpenFile(file_name));
return source;
}
bool RtpFileSource::ValidRtpDump(const std::string& file_name) {
std::unique_ptr<RtpFileReader> temp_file(
RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
return !!temp_file;
}
bool RtpFileSource::ValidPcap(const std::string& file_name) {
std::unique_ptr<RtpFileReader> temp_file(
RtpFileReader::Create(RtpFileReader::kPcap, file_name));
return !!temp_file;
}
RtpFileSource::~RtpFileSource() {}
bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
return rtp_header_extension_map_.RegisterByType(id, type);
}
std::unique_ptr<Packet> RtpFileSource::NextPacket() {
while (true) {
RtpPacket temp_packet;
if (!rtp_reader_->NextPacket(&temp_packet)) {
return NULL;
}
if (temp_packet.original_length == 0) {
// May be an RTCP packet.
// Read the next one.
continue;
}
std::unique_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]);
memcpy(packet_memory.get(), temp_packet.data, temp_packet.length);
RtpUtility::RtpHeaderParser parser(packet_memory.get(), temp_packet.length);
auto packet = std::make_unique<Packet>(
packet_memory.release(), temp_packet.length,
temp_packet.original_length, temp_packet.time_ms, parser,
&rtp_header_extension_map_);
if (!packet->valid_header()) {
continue;
}
if (filter_.test(packet->header().payloadType) ||
(ssrc_filter_ && packet->header().ssrc != *ssrc_filter_)) {
// This payload type should be filtered out. Continue to the next packet.
continue;
}
return packet;
}
}
RtpFileSource::RtpFileSource(absl::optional<uint32_t> ssrc_filter)
: PacketSource(),
ssrc_filter_(ssrc_filter) {}
bool RtpFileSource::OpenFile(const std::string& file_name) {
rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
if (rtp_reader_)
return true;
rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
if (!rtp_reader_) {
FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
"that .pcapng is not supported.";
}
return true;
}
} // namespace test
} // namespace webrtc