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