mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
c018bae807
git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^modules/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jared Siskin <jtsiskin@meta.com> Cr-Commit-Position: refs/heads/main@{#39901}
99 lines
3 KiB
C++
99 lines
3 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 <string.h>
|
|
|
|
#include "absl/strings/string_view.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(absl::string_view 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(absl::string_view file_name) {
|
|
std::unique_ptr<RtpFileReader> temp_file(
|
|
RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
|
|
return !!temp_file;
|
|
}
|
|
|
|
bool RtpFileSource::ValidPcap(absl::string_view 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;
|
|
}
|
|
auto packet = std::make_unique<Packet>(
|
|
rtc::CopyOnWriteBuffer(temp_packet.data, temp_packet.length),
|
|
temp_packet.original_length, temp_packet.time_ms,
|
|
&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(absl::string_view 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_) {
|
|
RTC_FATAL()
|
|
<< "Couldn't open input file as either a rtpdump or .pcap. Note "
|
|
<< "that .pcapng is not supported.";
|
|
}
|
|
return true;
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|