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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.
#include <stddef.h> // Provide access to size_t.
#include <map>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_headers.h"
#include "api/scoped_refptr.h"
namespace webrtc {
// Forward declarations.
class AudioFrame;
class AudioDecoderFactory;
class Clock;
struct NetEqNetworkStatistics {
uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
uint16_t preferred_buffer_size_ms; // Target buffer size in ms.
uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
// jitter; 0 otherwise.
uint16_t expand_rate; // Fraction (of original stream) of synthesized
// audio inserted through expansion (in Q14).
uint16_t speech_expand_rate; // Fraction (of original stream) of synthesized
// speech inserted through expansion (in Q14).
uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
// expansion (in Q14).
uint16_t accelerate_rate; // Fraction of data removed through acceleration
// (in Q14).
uint16_t secondary_decoded_rate; // Fraction of data coming from FEC/RED
// decoding (in Q14).
uint16_t secondary_discarded_rate; // Fraction of discarded FEC/RED data (in
// Q14).
// Statistics for packet waiting times, i.e., the time between a packet
// arrives until it is decoded.
int mean_waiting_time_ms;
int median_waiting_time_ms;
int min_waiting_time_ms;
int max_waiting_time_ms;
// NetEq statistics that persist over the lifetime of the class.
// These metrics are never reset.
struct NetEqLifetimeStatistics {
// Stats below correspond to similarly-named fields in the WebRTC stats spec.
uint64_t total_samples_received = 0;
uint64_t concealed_samples = 0;
uint64_t concealment_events = 0;
uint64_t jitter_buffer_delay_ms = 0;
uint64_t jitter_buffer_emitted_count = 0;
uint64_t jitter_buffer_target_delay_ms = 0;
uint64_t jitter_buffer_minimum_delay_ms = 0;
uint64_t inserted_samples_for_deceleration = 0;
uint64_t removed_samples_for_acceleration = 0;
uint64_t silent_concealed_samples = 0;
uint64_t fec_packets_received = 0;
uint64_t fec_packets_discarded = 0;
uint64_t packets_discarded = 0;
// Below stats are not part of the spec.
uint64_t delayed_packet_outage_samples = 0;
uint64_t delayed_packet_outage_events = 0;
// This is sum of relative packet arrival delays of received packets so far.
// Since end-to-end delay of a packet is difficult to measure and is not
// necessarily useful for measuring jitter buffer performance, we report a
// relative packet arrival delay. The relative packet arrival delay of a
// packet is defined as the arrival delay compared to the first packet
// received, given that it had zero delay. To avoid clock drift, the "first"
// packet can be made dynamic.
uint64_t relative_packet_arrival_delay_ms = 0;
uint64_t jitter_buffer_packets_received = 0;
// An interruption is a loss-concealment event lasting at least 150 ms. The
// two stats below count the number os such events and the total duration of
// these events.
int32_t interruption_count = 0;
int32_t total_interruption_duration_ms = 0;
// Total number of comfort noise samples generated during DTX.
uint64_t generated_noise_samples = 0;
// Metrics that describe the operations performed in NetEq, and the internal
// state.
struct NetEqOperationsAndState {
// These sample counters are cumulative, and don't reset. As a reference, the
// total number of output samples can be found in
// NetEqLifetimeStatistics::total_samples_received.
uint64_t preemptive_samples = 0;
uint64_t accelerate_samples = 0;
// Count of the number of buffer flushes.
uint64_t packet_buffer_flushes = 0;
// The statistics below are not cumulative.
// The waiting time of the last decoded packet.
uint64_t last_waiting_time_ms = 0;
// The sum of the packet and jitter buffer size in ms.
uint64_t current_buffer_size_ms = 0;
// The current frame size in ms.
uint64_t current_frame_size_ms = 0;
// Flag to indicate that the next packet is available.
bool next_packet_available = false;
// This is the interface class for NetEq.
class NetEq {
struct Config {
Config(const Config&);
Config& operator=(const Config&);
Config& operator=(Config&&);
std::string ToString() const;
int sample_rate_hz = 48000; // Initial value. Will change with input data.
size_t max_packets_in_buffer = 200;
int max_delay_ms = 0;
int min_delay_ms = 0;
bool enable_fast_accelerate = false;
bool enable_muted_state = false;
bool enable_rtx_handling = false;
absl::optional<AudioCodecPairId> codec_pair_id;
bool for_test_no_time_stretching = false; // Use only for testing.
enum ReturnCodes { kOK = 0, kFail = -1 };
enum class Operation {
enum class Mode {
// Return type for GetDecoderFormat.
struct DecoderFormat {
int sample_rate_hz;
int num_channels;
SdpAudioFormat sdp_format;
virtual ~NetEq() {}
// Inserts a new packet into NetEq.
// Returns 0 on success, -1 on failure.
virtual int InsertPacket(const RTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload) = 0;
// Lets NetEq know that a packet arrived with an empty payload. This typically
// happens when empty packets are used for probing the network channel, and
// these packets use RTP sequence numbers from the same series as the actual
// audio packets.
virtual void InsertEmptyPacket(const RTPHeader& rtp_header) = 0;
// Instructs NetEq to deliver 10 ms of audio data. The data is written to
// `audio_frame`. All data in `audio_frame` is wiped; `data_`, `speech_type_`,
// `num_channels_`, `sample_rate_hz_` and `samples_per_channel_` are updated
// upon success. If an error is returned, some fields may not have been
// updated, or may contain inconsistent values. If muted state is enabled
// (through Config::enable_muted_state), `muted` may be set to true after a
// prolonged expand period. When this happens, the `data_` in `audio_frame`
// is not written, but should be interpreted as being all zeros. For testing
// purposes, an override can be supplied in the `action_override` argument,
// which will cause NetEq to take this action next, instead of the action it
// would normally choose. An optional output argument for fetching the current
// sample rate can be provided, which will return the same value as
// last_output_sample_rate_hz() but will avoid additional synchronization.
// Returns kOK on success, or kFail in case of an error.
virtual int GetAudio(
AudioFrame* audio_frame,
bool* muted,
int* current_sample_rate_hz = nullptr,
absl::optional<Operation> action_override = absl::nullopt) = 0;
// Replaces the current set of decoders with the given one.
virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;
// Associates `rtp_payload_type` with the given codec, which NetEq will
// instantiate when it needs it. Returns true iff successful.
virtual bool RegisterPayloadType(int rtp_payload_type,
const SdpAudioFormat& audio_format) = 0;
// Removes `rtp_payload_type` from the codec database. Returns 0 on success,
// -1 on failure. Removing a payload type that is not registered is ok and
// will not result in an error.
virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
// Removes all payload types from the codec database.
virtual void RemoveAllPayloadTypes() = 0;
// Sets a minimum delay in millisecond for packet buffer. The minimum is
// maintained unless a higher latency is dictated by channel condition.
// Returns true if the minimum is successfully applied, otherwise false is
// returned.
virtual bool SetMinimumDelay(int delay_ms) = 0;
// Sets a maximum delay in milliseconds for packet buffer. The latency will
// not exceed the given value, even required delay (given the channel
// conditions) is higher. Calling this method has the same effect as setting
// the `max_delay_ms` value in the NetEq::Config struct.
virtual bool SetMaximumDelay(int delay_ms) = 0;
// Sets a base minimum delay in milliseconds for packet buffer. The minimum
// delay which is set via `SetMinimumDelay` can't be lower than base minimum
// delay. Calling this method is similar to setting the `min_delay_ms` value
// in the NetEq::Config struct. Returns true if the base minimum is
// successfully applied, otherwise false is returned.
virtual bool SetBaseMinimumDelayMs(int delay_ms) = 0;
// Returns current value of base minimum delay in milliseconds.
virtual int GetBaseMinimumDelayMs() const = 0;
// Returns the current target delay in ms. This includes any extra delay
// requested through SetMinimumDelay.
virtual int TargetDelayMs() const = 0;
// Returns the current total delay (packet buffer and sync buffer) in ms,
// with smoothing applied to even out short-time fluctuations due to jitter.
// The packet buffer part of the delay is not updated during DTX/CNG periods.
virtual int FilteredCurrentDelayMs() const = 0;
// Writes the current network statistics to `stats`. The statistics are reset
// after the call.
virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
// Current values only, not resetting any state.
virtual NetEqNetworkStatistics CurrentNetworkStatistics() const = 0;
// Returns a copy of this class's lifetime statistics. These statistics are
// never reset.
virtual NetEqLifetimeStatistics GetLifetimeStatistics() const = 0;
// Returns statistics about the performed operations and internal state. These
// statistics are never reset.
virtual NetEqOperationsAndState GetOperationsAndState() const = 0;
// Returns the RTP timestamp for the last sample delivered by GetAudio().
// The return value will be empty if no valid timestamp is available.
virtual absl::optional<uint32_t> GetPlayoutTimestamp() const = 0;
// Returns the sample rate in Hz of the audio produced in the last GetAudio
// call. If GetAudio has not been called yet, the configured sample rate
// (Config::sample_rate_hz) is returned.
virtual int last_output_sample_rate_hz() const = 0;
// Returns the decoder info for the given payload type. Returns empty if no
// such payload type was registered.
virtual absl::optional<DecoderFormat> GetDecoderFormat(
int payload_type) const = 0;
// Flushes both the packet buffer and the sync buffer.
virtual void FlushBuffers() = 0;
// Enables NACK and sets the maximum size of the NACK list, which should be
// positive and no larger than Nack::kNackListSizeLimit. If NACK is already
// enabled then the maximum NACK list size is modified accordingly.
virtual void EnableNack(size_t max_nack_list_size) = 0;
virtual void DisableNack() = 0;
// Returns a list of RTP sequence numbers corresponding to packets to be
// retransmitted, given an estimate of the round-trip time in milliseconds.
virtual std::vector<uint16_t> GetNackList(
int64_t round_trip_time_ms) const = 0;
// Returns the length of the audio yet to play in the sync buffer.
// Mainly intended for testing.
virtual int SyncBufferSizeMs() const = 0;
} // namespace webrtc
#endif // API_NETEQ_NETEQ_H_