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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.
*/
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
#include <algorithm>
#include <memory>
#include <string>
#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "rtc_base/buffer.h"
namespace webrtc {
namespace test {
// Interface class for input to the NetEqTest class.
class NetEqInput {
public:
struct PacketData {
PacketData();
~PacketData();
std::string ToString() const;
RTPHeader header;
rtc::Buffer payload;
int64_t time_ms;
};
virtual ~NetEqInput() = default;
// Returns at what time (in ms) NetEq::InsertPacket should be called next, or
// empty if the source is out of packets.
virtual absl::optional<int64_t> NextPacketTime() const = 0;
// Returns at what time (in ms) NetEq::GetAudio should be called next, or
// empty if no more output events are available.
virtual absl::optional<int64_t> NextOutputEventTime() const = 0;
// Returns the time (in ms) for the next event from either NextPacketTime()
// or NextOutputEventTime(), or empty if both are out of events.
absl::optional<int64_t> NextEventTime() const {
const auto a = NextPacketTime();
const auto b = NextOutputEventTime();
// Return the minimum of non-empty `a` and `b`, or empty if both are empty.
if (a) {
return b ? std::min(*a, *b) : a;
}
return b ? b : absl::nullopt;
}
// Returns the next packet to be inserted into NetEq. The packet following the
// returned one is pre-fetched in the NetEqInput object, such that future
// calls to NextPacketTime() or NextHeader() will return information from that
// packet.
virtual std::unique_ptr<PacketData> PopPacket() = 0;
// Move to the next output event. This will make NextOutputEventTime() return
// a new value (potentially the same if several output events share the same
// time).
virtual void AdvanceOutputEvent() = 0;
// Returns true if the source has come to an end. An implementation must
// eventually return true from this method, or the test will end up in an
// infinite loop.
virtual bool ended() const = 0;
// Returns the RTP header for the next packet, i.e., the packet that will be
// delivered next by PopPacket().
virtual absl::optional<RTPHeader> NextHeader() const = 0;
};
// Wrapper class to impose a time limit on a NetEqInput object, typically
// another time limit than what the object itself provides. For example, an
// input taken from a file can be cut shorter by wrapping it in this class.
class TimeLimitedNetEqInput : public NetEqInput {
public:
TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
~TimeLimitedNetEqInput() override;
absl::optional<int64_t> NextPacketTime() const override;
absl::optional<int64_t> NextOutputEventTime() const override;
std::unique_ptr<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
bool ended() const override;
absl::optional<RTPHeader> NextHeader() const override;
private:
void MaybeSetEnded();
std::unique_ptr<NetEqInput> input_;
const absl::optional<int64_t> start_time_ms_;
const int64_t duration_ms_;
bool ended_ = false;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
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