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/*
* 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_rtp_dump_input.h"
#include "absl/strings/string_view.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
namespace webrtc {
namespace test {
namespace {
// An adapter class to dress up a PacketSource object as a NetEqInput.
class NetEqRtpDumpInput : public NetEqInput {
public:
NetEqRtpDumpInput(absl::string_view file_name,
const std::map<int, RTPExtensionType>& 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> NextOutputEventTime() const override {
return next_output_event_ms_;
}
absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
return absl::nullopt;
}
void AdvanceOutputEvent() override {
if (next_output_event_ms_) {
*next_output_event_ms_ += kOutputPeriodMs;
}
if (!NextPacketTime()) {
next_output_event_ms_ = absl::nullopt;
}
}
void AdvanceSetMinimumDelay() override {}
absl::optional<int64_t> NextPacketTime() const override {
return packet_ ? absl::optional<int64_t>(
static_cast<int64_t>(packet_->time_ms()))
: absl::nullopt;
}
std::unique_ptr<PacketData> PopPacket() override {
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;
}
absl::optional<RTPHeader> NextHeader() const override {
return packet_ ? absl::optional<RTPHeader>(packet_->header())
: absl::nullopt;
}
bool ended() const override { return !next_output_event_ms_; }
private:
void LoadNextPacket() { packet_ = source_->NextPacket(); }
absl::optional<int64_t> next_output_event_ms_ = 0;
static constexpr int64_t kOutputPeriodMs = 10;
std::unique_ptr<RtpFileSource> source_;
std::unique_ptr<Packet> packet_;
};
} // namespace
std::unique_ptr<NetEqInput> CreateNetEqRtpDumpInput(
absl::string_view file_name,
const std::map<int, RTPExtensionType>& hdr_ext_map,
absl::optional<uint32_t> ssrc_filter) {
return std::make_unique<NetEqRtpDumpInput>(file_name, hdr_ext_map,
ssrc_filter);
}
} // namespace test
} // namespace webrtc