Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
This commit is contained in:
Arun Raghavan
257
webrtc/api/audio_codecs/audio_encoder.h
Normal file
257
webrtc/api/audio_codecs/audio_encoder.h
Normal file
@ -0,0 +1,257 @@
|
||||
/*
|
||||
* 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/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"
|
||||
#include "rtc_base/deprecation.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);
|
||||
|
||||
// This is to be deprecated. Please use |OnReceivedTargetAudioBitrate|
|
||||
// instead.
|
||||
// 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).
|
||||
RTC_DEPRECATED 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);
|
||||
|
||||
RTC_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;
|
||||
|
||||
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_
|
||||
Reference in New Issue
Block a user