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/*
* Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
#include "webrtc/api/optional.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class DecoderDatabase;
class StatisticsCalculator;
class TickTimer;
// This is the actual buffer holding the packets before decoding.
class PacketBuffer {
public:
enum BufferReturnCodes {
kOK = 0,
kFlushed,
kNotFound,
kBufferEmpty,
kInvalidPacket,
kInvalidPointer
};
// Constructor creates a buffer which can hold a maximum of
// |max_number_of_packets| packets.
PacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer);
// Deletes all packets in the buffer before destroying the buffer.
virtual ~PacketBuffer();
// Flushes the buffer and deletes all packets in it.
virtual void Flush();
// Returns true for an empty buffer.
virtual bool Empty() const;
// Inserts |packet| into the buffer. The buffer will take over ownership of
// the packet object.
// Returns PacketBuffer::kOK on success, PacketBuffer::kFlushed if the buffer
// was flushed due to overfilling.
virtual int InsertPacket(Packet&& packet, StatisticsCalculator* stats);
// Inserts a list of packets into the buffer. The buffer will take over
// ownership of the packet objects.
// Returns PacketBuffer::kOK if all packets were inserted successfully.
// If the buffer was flushed due to overfilling, only a subset of the list is
// inserted, and PacketBuffer::kFlushed is returned.
// The last three parameters are included for legacy compatibility.
// TODO(hlundin): Redesign to not use current_*_payload_type and
// decoder_database.
virtual int InsertPacketList(
PacketList* packet_list,
const DecoderDatabase& decoder_database,
rtc::Optional<uint8_t>* current_rtp_payload_type,
rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
StatisticsCalculator* stats);
// Gets the timestamp for the first packet in the buffer and writes it to the
// output variable |next_timestamp|.
// Returns PacketBuffer::kBufferEmpty if the buffer is empty,
// PacketBuffer::kOK otherwise.
virtual int NextTimestamp(uint32_t* next_timestamp) const;
// Gets the timestamp for the first packet in the buffer with a timestamp no
// lower than the input limit |timestamp|. The result is written to the output
// variable |next_timestamp|.
// Returns PacketBuffer::kBufferEmpty if the buffer is empty,
// PacketBuffer::kOK otherwise.
virtual int NextHigherTimestamp(uint32_t timestamp,
uint32_t* next_timestamp) const;
// Returns a (constant) pointer to the first packet in the buffer. Returns
// NULL if the buffer is empty.
virtual const Packet* PeekNextPacket() const;
// Extracts the first packet in the buffer and returns it.
// Returns an empty optional if the buffer is empty.
virtual rtc::Optional<Packet> GetNextPacket();
// Discards the first packet in the buffer. The packet is deleted.
// Returns PacketBuffer::kBufferEmpty if the buffer is empty,
// PacketBuffer::kOK otherwise.
virtual int DiscardNextPacket(StatisticsCalculator* stats);
// Discards all packets that are (strictly) older than timestamp_limit,
// but newer than timestamp_limit - horizon_samples. Setting horizon_samples
// to zero implies that the horizon is set to half the timestamp range. That
// is, if a packet is more than 2^31 timestamps into the future compared with
// timestamp_limit (including wrap-around), it is considered old.
virtual void DiscardOldPackets(uint32_t timestamp_limit,
uint32_t horizon_samples,
StatisticsCalculator* stats);
// Discards all packets that are (strictly) older than timestamp_limit.
virtual void DiscardAllOldPackets(uint32_t timestamp_limit,
StatisticsCalculator* stats);
// Removes all packets with a specific payload type from the buffer.
virtual void DiscardPacketsWithPayloadType(uint8_t payload_type,
StatisticsCalculator* stats);
// Returns the number of packets in the buffer, including duplicates and
// redundant packets.
virtual size_t NumPacketsInBuffer() const;
// Returns the number of samples in the buffer, including samples carried in
// duplicate and redundant packets.
virtual size_t NumSamplesInBuffer(size_t last_decoded_length) const;
virtual void BufferStat(int* num_packets, int* max_num_packets) const;
// Static method returning true if |timestamp| is older than |timestamp_limit|
// but less than |horizon_samples| behind |timestamp_limit|. For instance,
// with timestamp_limit = 100 and horizon_samples = 10, a timestamp in the
// range (90, 100) is considered obsolete, and will yield true.
// Setting |horizon_samples| to 0 is the same as setting it to 2^31, i.e.,
// half the 32-bit timestamp range.
static bool IsObsoleteTimestamp(uint32_t timestamp,
uint32_t timestamp_limit,
uint32_t horizon_samples) {
return IsNewerTimestamp(timestamp_limit, timestamp) &&
(horizon_samples == 0 ||
IsNewerTimestamp(timestamp, timestamp_limit - horizon_samples));
}
private:
size_t max_number_of_packets_;
PacketList buffer_;
const TickTimer* tick_timer_;
RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_