Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 260 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder.h b/third_party/libwebrtc/api/audio_codecs/audio_encoder.h
new file mode 100644
index 0000000000..7f5a34214f
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder.h
@@ -0,0 +1,260 @@
+/*
+ *  Copyright (c) 2014 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_AUDIO_CODECS_AUDIO_ENCODER_H_
+#define API_AUDIO_CODECS_AUDIO_ENCODER_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/call/bitrate_allocation.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+// Statistics related to Audio Network Adaptation.
+struct ANAStats {
+  ANAStats();
+  ANAStats(const ANAStats&);
+  ~ANAStats();
+  // Number of actions taken by the ANA bitrate controller since the start of
+  // the call. If this value is not set, it indicates that the bitrate
+  // controller is disabled.
+  absl::optional<uint32_t> bitrate_action_counter;
+  // Number of actions taken by the ANA channel controller since the start of
+  // the call. If this value is not set, it indicates that the channel
+  // controller is disabled.
+  absl::optional<uint32_t> channel_action_counter;
+  // Number of actions taken by the ANA DTX controller since the start of the
+  // call. If this value is not set, it indicates that the DTX controller is
+  // disabled.
+  absl::optional<uint32_t> dtx_action_counter;
+  // Number of actions taken by the ANA FEC controller since the start of the
+  // call. If this value is not set, it indicates that the FEC controller is
+  // disabled.
+  absl::optional<uint32_t> fec_action_counter;
+  // Number of times the ANA frame length controller decided to increase the
+  // frame length since the start of the call. If this value is not set, it
+  // indicates that the frame length controller is disabled.
+  absl::optional<uint32_t> frame_length_increase_counter;
+  // Number of times the ANA frame length controller decided to decrease the
+  // frame length since the start of the call. If this value is not set, it
+  // indicates that the frame length controller is disabled.
+  absl::optional<uint32_t> frame_length_decrease_counter;
+  // The uplink packet loss fractions as set by the ANA FEC controller. If this
+  // value is not set, it indicates that the ANA FEC controller is not active.
+  absl::optional<float> uplink_packet_loss_fraction;
+};
+
+// This is the interface class for encoders in AudioCoding module. Each codec
+// type must have an implementation of this class.
+class AudioEncoder {
+ public:
+  // Used for UMA logging of codec usage. The same codecs, with the
+  // same values, must be listed in
+  // src/tools/metrics/histograms/histograms.xml in chromium to log
+  // correct values.
+  enum class CodecType {
+    kOther = 0,  // Codec not specified, and/or not listed in this enum
+    kOpus = 1,
+    kIsac = 2,
+    kPcmA = 3,
+    kPcmU = 4,
+    kG722 = 5,
+    kIlbc = 6,
+
+    // Number of histogram bins in the UMA logging of codec types. The
+    // total number of different codecs that are logged cannot exceed this
+    // number.
+    kMaxLoggedAudioCodecTypes
+  };
+
+  struct EncodedInfoLeaf {
+    size_t encoded_bytes = 0;
+    uint32_t encoded_timestamp = 0;
+    int payload_type = 0;
+    bool send_even_if_empty = false;
+    bool speech = true;
+    CodecType encoder_type = CodecType::kOther;
+  };
+
+  // This is the main struct for auxiliary encoding information. Each encoded
+  // packet should be accompanied by one EncodedInfo struct, containing the
+  // total number of `encoded_bytes`, the `encoded_timestamp` and the
+  // `payload_type`. If the packet contains redundant encodings, the `redundant`
+  // vector will be populated with EncodedInfoLeaf structs. Each struct in the
+  // vector represents one encoding; the order of structs in the vector is the
+  // same as the order in which the actual payloads are written to the byte
+  // stream. When EncoderInfoLeaf structs are present in the vector, the main
+  // struct's `encoded_bytes` will be the sum of all the `encoded_bytes` in the
+  // vector.
+  struct EncodedInfo : public EncodedInfoLeaf {
+    EncodedInfo();
+    EncodedInfo(const EncodedInfo&);
+    EncodedInfo(EncodedInfo&&);
+    ~EncodedInfo();
+    EncodedInfo& operator=(const EncodedInfo&);
+    EncodedInfo& operator=(EncodedInfo&&);
+
+    std::vector<EncodedInfoLeaf> redundant;
+  };
+
+  virtual ~AudioEncoder() = default;
+
+  // Returns the input sample rate in Hz and the number of input channels.
+  // These are constants set at instantiation time.
+  virtual int SampleRateHz() const = 0;
+  virtual size_t NumChannels() const = 0;
+
+  // Returns the rate at which the RTP timestamps are updated. The default
+  // implementation returns SampleRateHz().
+  virtual int RtpTimestampRateHz() const;
+
+  // Returns the number of 10 ms frames the encoder will put in the next
+  // packet. This value may only change when Encode() outputs a packet; i.e.,
+  // the encoder may vary the number of 10 ms frames from packet to packet, but
+  // it must decide the length of the next packet no later than when outputting
+  // the preceding packet.
+  virtual size_t Num10MsFramesInNextPacket() const = 0;
+
+  // Returns the maximum value that can be returned by
+  // Num10MsFramesInNextPacket().
+  virtual size_t Max10MsFramesInAPacket() const = 0;
+
+  // Returns the current target bitrate in bits/s. The value -1 means that the
+  // codec adapts the target automatically, and a current target cannot be
+  // provided.
+  virtual int GetTargetBitrate() const = 0;
+
+  // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
+  // NumChannels() samples). Multi-channel audio must be sample-interleaved.
+  // The encoder appends zero or more bytes of output to `encoded` and returns
+  // additional encoding information.  Encode() checks some preconditions, calls
+  // EncodeImpl() which does the actual work, and then checks some
+  // postconditions.
+  EncodedInfo Encode(uint32_t rtp_timestamp,
+                     rtc::ArrayView<const int16_t> audio,
+                     rtc::Buffer* encoded);
+
+  // Resets the encoder to its starting state, discarding any input that has
+  // been fed to the encoder but not yet emitted in a packet.
+  virtual void Reset() = 0;
+
+  // Enables or disables codec-internal FEC (forward error correction). Returns
+  // true if the codec was able to comply. The default implementation returns
+  // true when asked to disable FEC and false when asked to enable it (meaning
+  // that FEC isn't supported).
+  virtual bool SetFec(bool enable);
+
+  // Enables or disables codec-internal VAD/DTX. Returns true if the codec was
+  // able to comply. The default implementation returns true when asked to
+  // disable DTX and false when asked to enable it (meaning that DTX isn't
+  // supported).
+  virtual bool SetDtx(bool enable);
+
+  // Returns the status of codec-internal DTX. The default implementation always
+  // returns false.
+  virtual bool GetDtx() const;
+
+  // Sets the application mode. Returns true if the codec was able to comply.
+  // The default implementation just returns false.
+  enum class Application { kSpeech, kAudio };
+  virtual bool SetApplication(Application application);
+
+  // Tells the encoder about the highest sample rate the decoder is expected to
+  // use when decoding the bitstream. The encoder would typically use this
+  // information to adjust the quality of the encoding. The default
+  // implementation does nothing.
+  virtual void SetMaxPlaybackRate(int frequency_hz);
+
+  // Tells the encoder what average bitrate we'd like it to produce. The
+  // encoder is free to adjust or disregard the given bitrate (the default
+  // implementation does the latter).
+  ABSL_DEPRECATED("Use OnReceivedTargetAudioBitrate instead")
+  virtual void SetTargetBitrate(int target_bps);
+
+  // Causes this encoder to let go of any other encoders it contains, and
+  // returns a pointer to an array where they are stored (which is required to
+  // live as long as this encoder). Unless the returned array is empty, you may
+  // not call any methods on this encoder afterwards, except for the
+  // destructor. The default implementation just returns an empty array.
+  // NOTE: This method is subject to change. Do not call or override it.
+  virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+  ReclaimContainedEncoders();
+
+  // Enables audio network adaptor. Returns true if successful.
+  virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                         RtcEventLog* event_log);
+
+  // Disables audio network adaptor.
+  virtual void DisableAudioNetworkAdaptor();
+
+  // Provides uplink packet loss fraction to this encoder to allow it to adapt.
+  // `uplink_packet_loss_fraction` is in the range [0.0, 1.0].
+  virtual void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction);
+
+  ABSL_DEPRECATED("")
+  virtual void OnReceivedUplinkRecoverablePacketLossFraction(
+      float uplink_recoverable_packet_loss_fraction);
+
+  // Provides target audio bitrate to this encoder to allow it to adapt.
+  virtual void OnReceivedTargetAudioBitrate(int target_bps);
+
+  // Provides target audio bitrate and corresponding probing interval of
+  // the bandwidth estimator to this encoder to allow it to adapt.
+  virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
+                                         absl::optional<int64_t> bwe_period_ms);
+
+  // Provides target audio bitrate and corresponding probing interval of
+  // the bandwidth estimator to this encoder to allow it to adapt.
+  virtual void OnReceivedUplinkAllocation(BitrateAllocationUpdate update);
+
+  // Provides RTT to this encoder to allow it to adapt.
+  virtual void OnReceivedRtt(int rtt_ms);
+
+  // Provides overhead to this encoder to adapt. The overhead is the number of
+  // bytes that will be added to each packet the encoder generates.
+  virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet);
+
+  // To allow encoder to adapt its frame length, it must be provided the frame
+  // length range that receivers can accept.
+  virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                           int max_frame_length_ms);
+
+  // Get statistics related to audio network adaptation.
+  virtual ANAStats GetANAStats() const;
+
+  // The range of frame lengths that are supported or nullopt if there's no sch
+  // information. This is used to calculated the full bitrate range, including
+  // overhead.
+  virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const = 0;
+
+  // The maximum number of audio channels supported by WebRTC encoders.
+  static constexpr int kMaxNumberOfChannels = 24;
+
+ protected:
+  // Subclasses implement this to perform the actual encoding. Called by
+  // Encode().
+  virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                                 rtc::ArrayView<const int16_t> audio,
+                                 rtc::Buffer* encoded) = 0;
+};
+}  // namespace webrtc
+#endif  // API_AUDIO_CODECS_AUDIO_ENCODER_H_