mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
714e3cbb48
Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
90 lines
2.9 KiB
C++
90 lines
2.9 KiB
C++
/*
|
|
* Copyright (c) 2016 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/neteq_packet_source_input.h"
|
|
|
|
#include <algorithm>
|
|
#include <limits>
|
|
|
|
#include "absl/strings/string_view.h"
|
|
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
|
|
NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {}
|
|
|
|
absl::optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
|
|
return packet_
|
|
? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
|
|
: absl::nullopt;
|
|
}
|
|
|
|
absl::optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
|
|
return packet_ ? absl::optional<RTPHeader>(packet_->header()) : absl::nullopt;
|
|
}
|
|
|
|
void NetEqPacketSourceInput::LoadNextPacket() {
|
|
packet_ = source()->NextPacket();
|
|
}
|
|
|
|
std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() {
|
|
if (!packet_) {
|
|
return std::unique_ptr<PacketData>();
|
|
}
|
|
std::unique_ptr<PacketData> packet_data(new PacketData);
|
|
packet_data->header = packet_->header();
|
|
if (packet_->payload_length_bytes() == 0 &&
|
|
packet_->virtual_payload_length_bytes() > 0) {
|
|
// This is a header-only "dummy" packet. Set the payload to all zeros, with
|
|
// length according to the virtual length.
|
|
packet_data->payload.SetSize(packet_->virtual_payload_length_bytes());
|
|
std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
|
|
} else {
|
|
packet_data->payload.SetData(packet_->payload(),
|
|
packet_->payload_length_bytes());
|
|
}
|
|
packet_data->time_ms = packet_->time_ms();
|
|
|
|
LoadNextPacket();
|
|
|
|
return packet_data;
|
|
}
|
|
|
|
NetEqRtpDumpInput::NetEqRtpDumpInput(absl::string_view file_name,
|
|
const RtpHeaderExtensionMap& hdr_ext_map,
|
|
absl::optional<uint32_t> ssrc_filter)
|
|
: source_(RtpFileSource::Create(file_name, ssrc_filter)) {
|
|
for (const auto& ext_pair : hdr_ext_map) {
|
|
source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
|
|
}
|
|
LoadNextPacket();
|
|
}
|
|
|
|
absl::optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
|
|
return next_output_event_ms_;
|
|
}
|
|
|
|
void NetEqRtpDumpInput::AdvanceOutputEvent() {
|
|
if (next_output_event_ms_) {
|
|
*next_output_event_ms_ += kOutputPeriodMs;
|
|
}
|
|
if (!NextPacketTime()) {
|
|
next_output_event_ms_ = absl::nullopt;
|
|
}
|
|
}
|
|
|
|
PacketSource* NetEqRtpDumpInput::source() {
|
|
return source_.get();
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|