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Remove some unused files

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This commit is contained in:
Arun Raghavan
2024-12-30 12:48:46 -05:00
parent 6119c05d7d
commit ecb8817972
30 changed files with 0 additions and 2295 deletions
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174
webrtc/api/audio_codecs/audio_decoder.cc
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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 <cstddef>
#include <cstdint>
#include <memory>
#include <optional>
#include <utility>
#include <vector>
#include "api/array_view.h"
#include "rtc_base/buffer.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);
}
std::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 ? std::nullopt
: std::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:
RTC_DCHECK_NOTREACHED();
return kSpeech;
}
}
constexpr int AudioDecoder::kMaxNumberOfChannels;
} // namespace webrtc

195
webrtc/api/audio_codecs/audio_decoder.h
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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 <optional>
#include <vector>
#include "api/array_view.h"
#include "rtc_base/buffer.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;
AudioDecoder(const AudioDecoder&) = delete;
AudioDecoder& operator=(const AudioDecoder&) = delete;
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 std::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
// std::optional. Decode may be called at most once per frame object.
virtual std::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 implementation.
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;
// The maximum number of audio channels supported by WebRTC decoders.
static constexpr int kMaxNumberOfChannels = 24;
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);
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_AUDIO_DECODER_H_

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webrtc/api/audio_codecs/audio_encoder.cc
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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 <cstddef>
#include <cstdint>
#include <memory>
#include <optional>
#include <string>
#include "api/array_view.h"
#include "api/call/bitrate_allocation.h"
#include "rtc_base/buffer.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_DCHECK_NOTREACHED();
}
void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
OnReceivedUplinkBandwidth(target_audio_bitrate_bps, std::nullopt);
}
void AudioEncoder::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
std::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();
}
constexpr int AudioEncoder::kMaxNumberOfChannels;
} // namespace webrtc

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webrtc/api/audio_codecs/audio_encoder.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.
*/
#ifndef API_AUDIO_CODECS_AUDIO_ENCODER_H_
#define API_AUDIO_CODECS_AUDIO_ENCODER_H_
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/base/attributes.h"
#include "api/array_view.h"
#include "api/call/bitrate_allocation.h"
#include "api/units/data_rate.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.
std::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.
std::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.
std::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.
std::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.
std::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.
std::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.
std::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,
std::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 such
// information. This is used together with the bitrate range to calculate the
// full bitrate range, including overhead.
virtual std::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const = 0;
// The range of payload bitrates that are supported. This is used together
// with the frame length range to calculate the full bitrate range, including
// overhead.
virtual std::optional<std::pair<DataRate, DataRate>> GetBitrateRange() const {
return std::nullopt;
}
// 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_

45
webrtc/api/call/bitrate_allocation.h
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@ -1,45 +0,0 @@
/*
* Copyright 2018 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_CALL_BITRATE_ALLOCATION_H_
#define API_CALL_BITRATE_ALLOCATION_H_
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
namespace webrtc {
// BitrateAllocationUpdate provides information to allocated streams about their
// bitrate allocation. It originates from the BitrateAllocater class and is
// propagated from there.
struct BitrateAllocationUpdate {
// The allocated target bitrate. Media streams should produce this amount of
// data. (Note that this may include packet overhead depending on
// configuration.)
DataRate target_bitrate = DataRate::Zero();
// The allocated part of the estimated link capacity. This is more stable than
// the target as it is based on the underlying link capacity estimate. This
// should be used to change encoder configuration when the cost of change is
// high.
DataRate stable_target_bitrate = DataRate::Zero();
// Predicted packet loss ratio.
double packet_loss_ratio = 0;
// Predicted round trip time.
TimeDelta round_trip_time = TimeDelta::PlusInfinity();
// `bwe_period` is deprecated, use `stable_target_bitrate` allocation instead.
TimeDelta bwe_period = TimeDelta::PlusInfinity();
// Congestion window pushback bitrate reduction fraction. Used in
// VideoStreamEncoder to reduce the bitrate by the given fraction
// by dropping frames.
double cwnd_reduce_ratio = 0;
};
} // namespace webrtc
#endif // API_CALL_BITRATE_ALLOCATION_H_

4
webrtc/api/meson.build
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@ -4,13 +4,9 @@ api_sources = [
'audio/audio_processing_statistics.cc',
'audio/channel_layout.cc',
'audio/echo_canceller3_config.cc',
'audio_codecs/audio_decoder.cc',
'audio_codecs/audio_encoder.cc',
'rtp_headers.cc',
'rtp_packet_info.cc',
'task_queue/task_queue_base.cc',
'units/data_rate.cc',
'units/data_size.cc',
'units/frequency.cc',
'units/time_delta.cc',
'units/timestamp.cc',

36
webrtc/api/units/data_rate.cc
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@ -1,36 +0,0 @@
/*
* Copyright (c) 2018 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/units/data_rate.h"
#include <string>
#include "api/array_view.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
std::string ToString(DataRate value) {
char buf[64];
rtc::SimpleStringBuilder sb(buf);
if (value.IsPlusInfinity()) {
sb << "+inf bps";
} else if (value.IsMinusInfinity()) {
sb << "-inf bps";
} else {
if (value.bps() == 0 || value.bps() % 1000 != 0) {
sb << value.bps() << " bps";
} else {
sb << value.kbps() << " kbps";
}
}
return sb.str();
}
} // namespace webrtc

153
webrtc/api/units/data_rate.h
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@ -1,153 +0,0 @@
/*
* Copyright (c) 2018 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_UNITS_DATA_RATE_H_
#define API_UNITS_DATA_RATE_H_
#include <cstdint>
#include <limits>
#include <string>
#include <type_traits>
#include "api/units/data_size.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "rtc_base/checks.h"
#include "rtc_base/system/rtc_export.h"
#include "rtc_base/units/unit_base.h" // IWYU pragma: export
namespace webrtc {
// DataRate is a class that represents a given data rate. This can be used to
// represent bandwidth, encoding bitrate, etc. The internal storage is bits per
// second (bps).
class DataRate final : public rtc_units_impl::RelativeUnit<DataRate> {
public:
template <typename T>
static constexpr DataRate BitsPerSec(T value) {
static_assert(std::is_arithmetic<T>::value, "");
return FromValue(value);
}
template <typename T>
static constexpr DataRate BytesPerSec(T value) {
static_assert(std::is_arithmetic<T>::value, "");
return FromFraction(8, value);
}
template <typename T>
static constexpr DataRate KilobitsPerSec(T value) {
static_assert(std::is_arithmetic<T>::value, "");
return FromFraction(1000, value);
}
static constexpr DataRate Infinity() { return PlusInfinity(); }
DataRate() = delete;
template <typename Sink>
friend void AbslStringify(Sink& sink, DataRate value);
template <typename T = int64_t>
constexpr T bps() const {
return ToValue<T>();
}
template <typename T = int64_t>
constexpr T bytes_per_sec() const {
return ToFraction<8, T>();
}
template <typename T = int64_t>
constexpr T kbps() const {
return ToFraction<1000, T>();
}
constexpr int64_t bps_or(int64_t fallback_value) const {
return ToValueOr(fallback_value);
}
constexpr int64_t kbps_or(int64_t fallback_value) const {
return ToFractionOr<1000>(fallback_value);
}
private:
// Bits per second used internally to simplify debugging by making the value
// more recognizable.
friend class rtc_units_impl::UnitBase<DataRate>;
using RelativeUnit::RelativeUnit;
static constexpr bool one_sided = true;
};
namespace data_rate_impl {
inline constexpr int64_t Microbits(const DataSize& size) {
constexpr int64_t kMaxBeforeConversion =
std::numeric_limits<int64_t>::max() / 8000000;
RTC_DCHECK_LE(size.bytes(), kMaxBeforeConversion)
<< "size is too large to be expressed in microbits";
return size.bytes() * 8000000;
}
inline constexpr int64_t MillibytePerSec(const DataRate& size) {
constexpr int64_t kMaxBeforeConversion =
std::numeric_limits<int64_t>::max() / (1000 / 8);
RTC_DCHECK_LE(size.bps(), kMaxBeforeConversion)
<< "rate is too large to be expressed in microbytes per second";
return size.bps() * (1000 / 8);
}
} // namespace data_rate_impl
inline constexpr DataRate operator/(const DataSize size,
const TimeDelta duration) {
return DataRate::BitsPerSec(data_rate_impl::Microbits(size) / duration.us());
}
inline constexpr TimeDelta operator/(const DataSize size, const DataRate rate) {
return TimeDelta::Micros(data_rate_impl::Microbits(size) / rate.bps());
}
inline constexpr DataSize operator*(const DataRate rate,
const TimeDelta duration) {
int64_t microbits = rate.bps() * duration.us();
return DataSize::Bytes((microbits + 4000000) / 8000000);
}
inline constexpr DataSize operator*(const TimeDelta duration,
const DataRate rate) {
return rate * duration;
}
inline constexpr DataSize operator/(const DataRate rate,
const Frequency frequency) {
int64_t millihertz = frequency.millihertz<int64_t>();
// Note that the value is truncated here reather than rounded, potentially
// introducing an error of .5 bytes if rounding were expected.
return DataSize::Bytes(data_rate_impl::MillibytePerSec(rate) / millihertz);
}
inline constexpr Frequency operator/(const DataRate rate, const DataSize size) {
return Frequency::MilliHertz(data_rate_impl::MillibytePerSec(rate) /
size.bytes());
}
inline constexpr DataRate operator*(const DataSize size,
const Frequency frequency) {
RTC_DCHECK(frequency.IsZero() ||
size.bytes() <= std::numeric_limits<int64_t>::max() / 8 /
frequency.millihertz<int64_t>());
int64_t millibits_per_second =
size.bytes() * 8 * frequency.millihertz<int64_t>();
return DataRate::BitsPerSec((millibits_per_second + 500) / 1000);
}
inline constexpr DataRate operator*(const Frequency frequency,
const DataSize size) {
return size * frequency;
}
RTC_EXPORT std::string ToString(DataRate value);
inline std::string ToLogString(DataRate value) {
return ToString(value);
}
template <typename Sink>
void AbslStringify(Sink& sink, DataRate value) {
sink.Append(ToString(value));
}
} // namespace webrtc
#endif // API_UNITS_DATA_RATE_H_

32
webrtc/api/units/data_size.cc
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@ -1,32 +0,0 @@
/*
* Copyright (c) 2018 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/units/data_size.h"
#include <string>
#include "api/array_view.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
std::string ToString(DataSize value) {
char buf[64];
rtc::SimpleStringBuilder sb(buf);
if (value.IsPlusInfinity()) {
sb << "+inf bytes";
} else if (value.IsMinusInfinity()) {
sb << "-inf bytes";
} else {
sb << value.bytes() << " bytes";
}
return sb.str();
}
} // namespace webrtc

64
webrtc/api/units/data_size.h
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@ -1,64 +0,0 @@
/*
* Copyright (c) 2018 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_UNITS_DATA_SIZE_H_
#define API_UNITS_DATA_SIZE_H_
#include <cstdint>
#include <string>
#include <type_traits>
#include "rtc_base/system/rtc_export.h"
#include "rtc_base/units/unit_base.h" // IWYU pragma: export
namespace webrtc {
// DataSize is a class represeting a count of bytes.
class DataSize final : public rtc_units_impl::RelativeUnit<DataSize> {
public:
template <typename T>
static constexpr DataSize Bytes(T value) {
static_assert(std::is_arithmetic<T>::value, "");
return FromValue(value);
}
static constexpr DataSize Infinity() { return PlusInfinity(); }
DataSize() = delete;
template <typename Sink>
friend void AbslStringify(Sink& sink, DataSize value);
template <typename T = int64_t>
constexpr T bytes() const {
return ToValue<T>();
}
constexpr int64_t bytes_or(int64_t fallback_value) const {
return ToValueOr(fallback_value);
}
private:
friend class rtc_units_impl::UnitBase<DataSize>;
using RelativeUnit::RelativeUnit;
static constexpr bool one_sided = true;
};
RTC_EXPORT std::string ToString(DataSize value);
inline std::string ToLogString(DataSize value) {
return ToString(value);
}
template <typename Sink>
void AbslStringify(Sink& sink, DataSize value) {
sink.Append(ToString(value));
}
} // namespace webrtc
#endif // API_UNITS_DATA_SIZE_H_