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* 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.
#include <algorithm>
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/api/optional.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/deprecation.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class Clock;
class RtcEventLog;
// This is the interface class for encoders in AudioCoding module. Each codec
// type must have an implementation of this class.
class AudioEncoder {
// 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.
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(const 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>>
// 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);
// Provides 1st-order-FEC-recoverable uplink packet loss rate to this encoder
// to allow it to adapt.
// |uplink_recoverable_packet_loss_fraction| is in the range [0.0, 1.0].
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,
rtc::Optional<int64_t> bwe_period_ms);
// 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);
// 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