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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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170
webrtc/api/audio_codecs/audio_decoder.cc Normal file
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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.
*/
#include "api/audio_codecs/audio_decoder.h"
#include <assert.h>
#include <memory>
#include <utility>
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/sanitizer.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
public:
OldStyleEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload)
: decoder_(decoder), payload_(std::move(payload)) {}
size_t Duration() const override {
const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
return ret < 0 ? 0 : static_cast<size_t>(ret);
}
absl::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const override {
auto speech_type = AudioDecoder::kSpeech;
const int ret = decoder_->Decode(
payload_.data(), payload_.size(), decoder_->SampleRateHz(),
decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
return ret < 0 ? absl::nullopt
: absl::optional<DecodeResult>(
{static_cast<size_t>(ret), speech_type});
}
private:
AudioDecoder* const decoder_;
const rtc::Buffer payload_;
};
} // namespace
bool AudioDecoder::EncodedAudioFrame::IsDtxPacket() const {
return false;
}
AudioDecoder::ParseResult::ParseResult() = default;
AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
int priority,
std::unique_ptr<EncodedAudioFrame> frame)
: timestamp(timestamp), priority(priority), frame(std::move(frame)) {
RTC_DCHECK_GE(priority, 0);
}
AudioDecoder::ParseResult::~ParseResult() = default;
AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=(
ParseResult&& b) = default;
std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
std::vector<ParseResult> results;
std::unique_ptr<EncodedAudioFrame> frame(
new OldStyleEncodedFrame(this, std::move(payload)));
results.emplace_back(timestamp, 0, std::move(frame));
return results;
}
int AudioDecoder::Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type) {
TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
int duration = PacketDuration(encoded, encoded_len);
if (duration >= 0 &&
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
return -1;
}
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
speech_type);
}
int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type) {
TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
int duration = PacketDurationRedundant(encoded, encoded_len);
if (duration >= 0 &&
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
return -1;
}
return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
speech_type);
}
int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
speech_type);
}
bool AudioDecoder::HasDecodePlc() const {
return false;
}
size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
return 0;
}
// TODO(bugs.webrtc.org/9676): Remove default implementation.
void AudioDecoder::GeneratePlc(size_t /*requested_samples_per_channel*/,
rtc::BufferT<int16_t>* /*concealment_audio*/) {}
int AudioDecoder::ErrorCode() {
return 0;
}
int AudioDecoder::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
return kNotImplemented;
}
int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
return kNotImplemented;
}
bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
size_t encoded_len) const {
return false;
}
AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
switch (type) {
case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech.
case 1:
return kSpeech;
case 2:
return kComfortNoise;
default:
assert(false);
return kSpeech;
}
}
} // namespace webrtc

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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_AUDIO_CODECS_AUDIO_DECODER_H_
#define API_AUDIO_CODECS_AUDIO_DECODER_H_
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/buffer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class AudioDecoder {
public:
enum SpeechType {
kSpeech = 1,
kComfortNoise = 2,
};
// Used by PacketDuration below. Save the value -1 for errors.
enum { kNotImplemented = -2 };
AudioDecoder() = default;
virtual ~AudioDecoder() = default;
class EncodedAudioFrame {
public:
struct DecodeResult {
size_t num_decoded_samples;
SpeechType speech_type;
};
virtual ~EncodedAudioFrame() = default;
// Returns the duration in samples-per-channel of this audio frame.
// If no duration can be ascertained, returns zero.
virtual size_t Duration() const = 0;
// Returns true if this packet contains DTX.
virtual bool IsDtxPacket() const;
// Decodes this frame of audio and writes the result in |decoded|.
// |decoded| must be large enough to store as many samples as indicated by a
// call to Duration() . On success, returns an absl::optional containing the
// total number of samples across all channels, as well as whether the
// decoder produced comfort noise or speech. On failure, returns an empty
// absl::optional. Decode may be called at most once per frame object.
virtual absl::optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const = 0;
};
struct ParseResult {
ParseResult();
ParseResult(uint32_t timestamp,
int priority,
std::unique_ptr<EncodedAudioFrame> frame);
ParseResult(ParseResult&& b);
~ParseResult();
ParseResult& operator=(ParseResult&& b);
// The timestamp of the frame is in samples per channel.
uint32_t timestamp;
// The relative priority of the frame compared to other frames of the same
// payload and the same timeframe. A higher value means a lower priority.
// The highest priority is zero - negative values are not allowed.
int priority;
std::unique_ptr<EncodedAudioFrame> frame;
};
// Let the decoder parse this payload and prepare zero or more decodable
// frames. Each frame must be between 10 ms and 120 ms long. The caller must
// ensure that the AudioDecoder object outlives any frame objects returned by
// this call. The decoder is free to swap or move the data from the |payload|
// buffer. |timestamp| is the input timestamp, in samples, corresponding to
// the start of the payload.
virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
uint32_t timestamp);
// TODO(bugs.webrtc.org/10098): The Decode and DecodeRedundant methods are
// obsolete; callers should call ParsePayload instead. For now, subclasses
// must still implement DecodeInternal.
// Decodes |encode_len| bytes from |encoded| and writes the result in
// |decoded|. The maximum bytes allowed to be written into |decoded| is
// |max_decoded_bytes|. Returns the total number of samples across all
// channels. If the decoder produced comfort noise, |speech_type|
// is set to kComfortNoise, otherwise it is kSpeech. The desired output
// sample rate is provided in |sample_rate_hz|, which must be valid for the
// codec at hand.
int Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type);
// Same as Decode(), but interfaces to the decoders redundant decode function.
// The default implementation simply calls the regular Decode() method.
int DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type);
// Indicates if the decoder implements the DecodePlc method.
virtual bool HasDecodePlc() const;
// Calls the packet-loss concealment of the decoder to update the state after
// one or several lost packets. The caller has to make sure that the
// memory allocated in |decoded| should accommodate |num_frames| frames.
virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
// Asks the decoder to generate packet-loss concealment and append it to the
// end of |concealment_audio|. The concealment audio should be in
// channel-interleaved format, with as many channels as the last decoded
// packet produced. The implementation must produce at least
// requested_samples_per_channel, or nothing at all. This is a signal to the
// caller to conceal the loss with other means. If the implementation provides
// concealment samples, it is also responsible for "stitching" it together
// with the decoded audio on either side of the concealment.
// Note: The default implementation of GeneratePlc will be deleted soon. All
// implementations must provide their own, which can be a simple as a no-op.
// TODO(bugs.webrtc.org/9676): Remove default impementation.
virtual void GeneratePlc(size_t requested_samples_per_channel,
rtc::BufferT<int16_t>* concealment_audio);
// Resets the decoder state (empty buffers etc.).
virtual void Reset() = 0;
// Returns the last error code from the decoder.
virtual int ErrorCode();
// Returns the duration in samples-per-channel of the payload in |encoded|
// which is |encoded_len| bytes long. Returns kNotImplemented if no duration
// estimate is available, or -1 in case of an error.
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
// Returns the duration in samples-per-channel of the redandant payload in
// |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
// duration estimate is available, or -1 in case of an error.
virtual int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const;
// Detects whether a packet has forward error correction. The packet is
// comprised of the samples in |encoded| which is |encoded_len| bytes long.
// Returns true if the packet has FEC and false otherwise.
virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
// Returns the actual sample rate of the decoder's output. This value may not
// change during the lifetime of the decoder.
virtual int SampleRateHz() const = 0;
// The number of channels in the decoder's output. This value may not change
// during the lifetime of the decoder.
virtual size_t Channels() const = 0;
protected:
static SpeechType ConvertSpeechType(int16_t type);
virtual int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) = 0;
virtual int DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_AUDIO_DECODER_H_

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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 "api/audio_codecs/audio_encoder.h"
#include "rtc_base/checks.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
ANAStats::ANAStats() = default;
ANAStats::~ANAStats() = default;
ANAStats::ANAStats(const ANAStats&) = default;
AudioEncoder::EncodedInfo::EncodedInfo() = default;
AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
AudioEncoder::EncodedInfo::~EncodedInfo() = default;
AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
const EncodedInfo&) = default;
AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
default;
int AudioEncoder::RtpTimestampRateHz() const {
return SampleRateHz();
}
AudioEncoder::EncodedInfo AudioEncoder::Encode(
uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) {
TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
RTC_CHECK_EQ(audio.size(),
static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
const size_t old_size = encoded->size();
EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
return info;
}
bool AudioEncoder::SetFec(bool enable) {
return !enable;
}
bool AudioEncoder::SetDtx(bool enable) {
return !enable;
}
bool AudioEncoder::GetDtx() const {
return false;
}
bool AudioEncoder::SetApplication(Application application) {
return false;
}
void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
void AudioEncoder::SetTargetBitrate(int target_bps) {}
rtc::ArrayView<std::unique_ptr<AudioEncoder>>
AudioEncoder::ReclaimContainedEncoders() {
return nullptr;
}
bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
RtcEventLog* event_log) {
return false;
}
void AudioEncoder::DisableAudioNetworkAdaptor() {}
void AudioEncoder::OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) {}
void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
float uplink_recoverable_packet_loss_fraction) {
RTC_NOTREACHED();
}
void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
}
void AudioEncoder::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
absl::optional<int64_t> bwe_period_ms) {}
void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
update.bwe_period.ms());
}
void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) {}
ANAStats AudioEncoder::GetANAStats() const {
return ANAStats();
}
} // namespace webrtc

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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.
*/
#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_