Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 322 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/neteq/neteq.h b/third_party/libwebrtc/api/neteq/neteq.h
new file mode 100644
index 0000000000..5300c5601e
--- /dev/null
+++ b/third_party/libwebrtc/api/neteq/neteq.h
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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 API_NETEQ_NETEQ_H_
+#define API_NETEQ_NETEQ_H_
+
+#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.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
+  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;
+  // 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 {
+ public:
+  struct Config {
+    Config();
+    Config(const Config&);
+    Config(Config&&);
+    ~Config();
+    Config& operator=(const Config&);
+    Config& operator=(Config&&);
+
+    std::string ToString() const;
+
+    int sample_rate_hz = 48000;  // Initial value. Will change with input data.
+    bool enable_post_decode_vad = false;
+    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 {
+    kNormal,
+    kMerge,
+    kExpand,
+    kAccelerate,
+    kFastAccelerate,
+    kPreemptiveExpand,
+    kRfc3389Cng,
+    kRfc3389CngNoPacket,
+    kCodecInternalCng,
+    kDtmf,
+    kUndefined,
+  };
+
+  enum class Mode {
+    kNormal,
+    kExpand,
+    kMerge,
+    kAccelerateSuccess,
+    kAccelerateLowEnergy,
+    kAccelerateFail,
+    kPreemptiveExpandSuccess,
+    kPreemptiveExpandLowEnergy,
+    kPreemptiveExpandFail,
+    kRfc3389Cng,
+    kCodecInternalCng,
+    kCodecPlc,
+    kDtmf,
+    kError,
+    kUndefined,
+  };
+
+  // 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_`, `samples_per_channel_`, and
+  // `vad_activity_` 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;
+
+  // Enables post-decode VAD. When enabled, GetAudio() will return
+  // kOutputVADPassive when the signal contains no speech.
+  virtual void EnableVad() = 0;
+
+  // Disables post-decode VAD.
+  virtual void DisableVad() = 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_