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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_

3
webrtc/common_audio/meson.build
View File

@ -4,8 +4,6 @@ common_audio_sources = [
'channel_buffer.cc',
'fir_filter_c.cc',
'fir_filter_factory.cc',
'real_fourier.cc',
'real_fourier_ooura.cc',
'resampler/push_resampler.cc',
'resampler/push_sinc_resampler.cc',
'resampler/resampler.cc',
@ -55,7 +53,6 @@ common_audio_sources = [
'vad/vad_gmm.c',
'vad/vad_sp.c',
'vad/webrtc_vad.c',
'window_generator.cc',
]
arch_libs = []

51
webrtc/common_audio/real_fourier.cc
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@ -1,51 +0,0 @@
/*
* 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 "common_audio/real_fourier.h"
#include "common_audio/real_fourier_ooura.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "rtc_base/checks.h"
namespace webrtc {
using std::complex;
const size_t RealFourier::kFftBufferAlignment = 32;
std::unique_ptr<RealFourier> RealFourier::Create(int fft_order) {
return std::unique_ptr<RealFourier>(new RealFourierOoura(fft_order));
}
int RealFourier::FftOrder(size_t length) {
RTC_CHECK_GT(length, 0U);
return WebRtcSpl_GetSizeInBits(static_cast<uint32_t>(length - 1));
}
size_t RealFourier::FftLength(int order) {
RTC_CHECK_GE(order, 0);
return size_t{1} << order;
}
size_t RealFourier::ComplexLength(int order) {
return FftLength(order) / 2 + 1;
}
RealFourier::fft_real_scoper RealFourier::AllocRealBuffer(int count) {
return fft_real_scoper(static_cast<float*>(
AlignedMalloc(sizeof(float) * count, kFftBufferAlignment)));
}
RealFourier::fft_cplx_scoper RealFourier::AllocCplxBuffer(int count) {
return fft_cplx_scoper(static_cast<complex<float>*>(
AlignedMalloc(sizeof(complex<float>) * count, kFftBufferAlignment)));
}
} // namespace webrtc

76
webrtc/common_audio/real_fourier.h
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@ -1,76 +0,0 @@
/*
* 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 COMMON_AUDIO_REAL_FOURIER_H_
#define COMMON_AUDIO_REAL_FOURIER_H_
#include <stddef.h>
#include <complex>
#include <memory>
#include "rtc_base/memory/aligned_malloc.h"
// Uniform interface class for the real DFT and its inverse, for power-of-2
// input lengths. Also contains helper functions for buffer allocation, taking
// care of any memory alignment requirements the underlying library might have.
namespace webrtc {
class RealFourier {
public:
// Shorthand typenames for the scopers used by the buffer allocation helpers.
typedef std::unique_ptr<float[], AlignedFreeDeleter> fft_real_scoper;
typedef std::unique_ptr<std::complex<float>[], AlignedFreeDeleter>
fft_cplx_scoper;
// The alignment required for all input and output buffers, in bytes.
static const size_t kFftBufferAlignment;
// Construct a wrapper instance for the given input order, which must be
// between 1 and kMaxFftOrder, inclusively.
static std::unique_ptr<RealFourier> Create(int fft_order);
virtual ~RealFourier() {}
// Helper to compute the smallest FFT order (a power of 2) which will contain
// the given input length.
static int FftOrder(size_t length);
// Helper to compute the input length from the FFT order.
static size_t FftLength(int order);
// Helper to compute the exact length, in complex floats, of the transform
// output (i.e. |2^order / 2 + 1|).
static size_t ComplexLength(int order);
// Buffer allocation helpers. The buffers are large enough to hold `count`
// floats/complexes and suitably aligned for use by the implementation.
// The returned scopers are set up with proper deleters; the caller owns
// the allocated memory.
static fft_real_scoper AllocRealBuffer(int count);
static fft_cplx_scoper AllocCplxBuffer(int count);
// Main forward transform interface. The output array need only be big
// enough for |2^order / 2 + 1| elements - the conjugate pairs are not
// returned. Input and output must be properly aligned (e.g. through
// AllocRealBuffer and AllocCplxBuffer) and input length must be
// |2^order| (same as given at construction time).
virtual void Forward(const float* src, std::complex<float>* dest) const = 0;
// Inverse transform. Same input format as output above, conjugate pairs
// not needed.
virtual void Inverse(const std::complex<float>* src, float* dest) const = 0;
virtual int order() const = 0;
};
} // namespace webrtc
#endif // COMMON_AUDIO_REAL_FOURIER_H_

91
webrtc/common_audio/real_fourier_ooura.cc
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@ -1,91 +0,0 @@
/*
* Copyright (c) 2015 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 "common_audio/real_fourier_ooura.h"
#include <algorithm>
#include <cmath>
#include "common_audio/third_party/ooura/fft_size_256/fft4g.h"
#include "rtc_base/checks.h"
namespace webrtc {
using std::complex;
namespace {
void Conjugate(complex<float>* array, size_t complex_length) {
std::for_each(array, array + complex_length,
[=](complex<float>& v) { v = std::conj(v); });
}
size_t ComputeWorkIpSize(size_t fft_length) {
return static_cast<size_t>(
2 + std::ceil(std::sqrt(static_cast<float>(fft_length))));
}
} // namespace
RealFourierOoura::RealFourierOoura(int fft_order)
: order_(fft_order),
length_(FftLength(order_)),
complex_length_(ComplexLength(order_)),
// Zero-initializing work_ip_ will cause rdft to initialize these work
// arrays on the first call.
work_ip_(new size_t[ComputeWorkIpSize(length_)]()),
work_w_(new float[complex_length_]()) {
RTC_CHECK_GE(fft_order, 1);
}
RealFourierOoura::~RealFourierOoura() = default;
void RealFourierOoura::Forward(const float* src, complex<float>* dest) const {
{
// This cast is well-defined since C++11. See "Non-static data members" at:
// http://en.cppreference.com/w/cpp/numeric/complex
auto* dest_float = reinterpret_cast<float*>(dest);
std::copy(src, src + length_, dest_float);
WebRtc_rdft(length_, 1, dest_float, work_ip_.get(), work_w_.get());
}
// Ooura places real[n/2] in imag[0].
dest[complex_length_ - 1] = complex<float>(dest[0].imag(), 0.0f);
dest[0] = complex<float>(dest[0].real(), 0.0f);
// Ooura returns the conjugate of the usual Fourier definition.
Conjugate(dest, complex_length_);
}
void RealFourierOoura::Inverse(const complex<float>* src, float* dest) const {
{
auto* dest_complex = reinterpret_cast<complex<float>*>(dest);
// The real output array is shorter than the input complex array by one
// complex element.
const size_t dest_complex_length = complex_length_ - 1;
std::copy(src, src + dest_complex_length, dest_complex);
// Restore Ooura's conjugate definition.
Conjugate(dest_complex, dest_complex_length);
// Restore real[n/2] to imag[0].
dest_complex[0] =
complex<float>(dest_complex[0].real(), src[complex_length_ - 1].real());
}
WebRtc_rdft(length_, -1, dest, work_ip_.get(), work_w_.get());
// Ooura returns a scaled version.
const float scale = 2.0f / length_;
std::for_each(dest, dest + length_, [scale](float& v) { v *= scale; });
}
int RealFourierOoura::order() const {
return order_;
}
} // namespace webrtc

45
webrtc/common_audio/real_fourier_ooura.h
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@ -1,45 +0,0 @@
/*
* Copyright (c) 2015 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 COMMON_AUDIO_REAL_FOURIER_OOURA_H_
#define COMMON_AUDIO_REAL_FOURIER_OOURA_H_
#include <stddef.h>
#include <complex>
#include <memory>
#include "common_audio/real_fourier.h"
namespace webrtc {
class RealFourierOoura : public RealFourier {
public:
explicit RealFourierOoura(int fft_order);
~RealFourierOoura() override;
void Forward(const float* src, std::complex<float>* dest) const override;
void Inverse(const std::complex<float>* src, float* dest) const override;
int order() const override;
private:
const int order_;
const size_t length_;
const size_t complex_length_;
// These are work arrays for Ooura. The names are based on the comments in
// common_audio/third_party/ooura/fft_size_256/fft4g.cc.
const std::unique_ptr<size_t[]> work_ip_;
const std::unique_ptr<float[]> work_w_;
};
} // namespace webrtc
#endif // COMMON_AUDIO_REAL_FOURIER_OOURA_H_

44
webrtc/common_audio/real_fourier_openmax.h
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@ -1,44 +0,0 @@
/*
* Copyright (c) 2015 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 WEBRTC_COMMON_AUDIO_REAL_FOURIER_OPENMAX_H_
#define WEBRTC_COMMON_AUDIO_REAL_FOURIER_OPENMAX_H_
#include <complex>
#include "webrtc/common_audio/real_fourier.h"
namespace webrtc {
class RealFourierOpenmax : public RealFourier {
public:
explicit RealFourierOpenmax(int fft_order);
~RealFourierOpenmax() override;
void Forward(const float* src, std::complex<float>* dest) const override;
void Inverse(const std::complex<float>* src, float* dest) const override;
int order() const override {
return order_;
}
private:
// Basically a forward declare of OMXFFTSpec_R_F32. To get rid of the
// dependency on openmax.
typedef void OMXFFTSpec_R_F32_;
const int order_;
OMXFFTSpec_R_F32_* const omx_spec_;
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_REAL_FOURIER_OPENMAX_H_

71
webrtc/common_audio/window_generator.cc
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@ -1,71 +0,0 @@
/*
* 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.
*/
#define _USE_MATH_DEFINES
#include "common_audio/window_generator.h"
#include <cmath>
#include <complex>
#include "rtc_base/checks.h"
using std::complex;
namespace {
// Modified Bessel function of order 0 for complex inputs.
complex<float> I0(complex<float> x) {
complex<float> y = x / 3.75f;
y *= y;
return 1.0f + y * (3.5156229f +
y * (3.0899424f +
y * (1.2067492f +
y * (0.2659732f +
y * (0.360768e-1f + y * 0.45813e-2f)))));
}
} // namespace
namespace webrtc {
void WindowGenerator::Hanning(int length, float* window) {
RTC_CHECK_GT(length, 1);
RTC_CHECK(window != nullptr);
for (int i = 0; i < length; ++i) {
window[i] =
0.5f * (1 - cosf(2 * static_cast<float>(M_PI) * i / (length - 1)));
}
}
void WindowGenerator::KaiserBesselDerived(float alpha,
size_t length,
float* window) {
RTC_CHECK_GT(length, 1U);
RTC_CHECK(window != nullptr);
const size_t half = (length + 1) / 2;
float sum = 0.0f;
for (size_t i = 0; i <= half; ++i) {
complex<float> r = (4.0f * i) / length - 1.0f;
sum += I0(static_cast<float>(M_PI) * alpha * sqrt(1.0f - r * r)).real();
window[i] = sum;
}
for (size_t i = length - 1; i >= half; --i) {
window[length - i - 1] = sqrtf(window[length - i - 1] / sum);
window[i] = window[length - i - 1];
}
if (length % 2 == 1) {
window[half - 1] = sqrtf(window[half - 1] / sum);
}
}
} // namespace webrtc

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webrtc/common_audio/window_generator.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 COMMON_AUDIO_WINDOW_GENERATOR_H_
#define COMMON_AUDIO_WINDOW_GENERATOR_H_
#include <stddef.h>
namespace webrtc {
// Helper class with generators for various signal transform windows.
class WindowGenerator {
public:
WindowGenerator() = delete;
WindowGenerator(const WindowGenerator&) = delete;
WindowGenerator& operator=(const WindowGenerator&) = delete;
static void Hanning(int length, float* window);
static void KaiserBesselDerived(float alpha, size_t length, float* window);
};
} // namespace webrtc
#endif // COMMON_AUDIO_WINDOW_GENERATOR_H_

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webrtc/modules/audio_coding/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.
*/
// This file is for backwards compatibility only! Use
// webrtc/api/audio_codecs/audio_decoder.h instead!
// TODO(kwiberg): Remove it.
#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
#define MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
#include "api/audio_codecs/audio_decoder.h"
#endif // MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_

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webrtc/modules/audio_coding/codecs/audio_encoder.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.
*/
// This file is for backwards compatibility only! Use
// webrtc/api/audio_codecs/audio_encoder.h instead!
// TODO(ossu): Remove it.
#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
#define MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
#include "api/audio_codecs/audio_encoder.h"
#endif // MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_

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webrtc/modules/audio_coding/codecs/cng/webrtc_cng.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 WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INCLUDE_WEBRTC_CNG_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INCLUDE_WEBRTC_CNG_H_
#include <stddef.h>
#include "webrtc/typedefs.h"
#ifdef __cplusplus
extern "C" {
#endif
#define WEBRTC_CNG_MAX_LPC_ORDER 12
#define WEBRTC_CNG_MAX_OUTSIZE_ORDER 640
/* Define Error codes. */
/* 6100 Encoder */
#define CNG_ENCODER_NOT_INITIATED 6120
#define CNG_DISALLOWED_LPC_ORDER 6130
#define CNG_DISALLOWED_FRAME_SIZE 6140
#define CNG_DISALLOWED_SAMPLING_FREQUENCY 6150
/* 6200 Decoder */
#define CNG_DECODER_NOT_INITIATED 6220
typedef struct WebRtcCngEncInst CNG_enc_inst;
typedef struct WebRtcCngDecInst CNG_dec_inst;
/****************************************************************************
* WebRtcCng_CreateEnc/Dec(...)
*
* These functions create an instance to the specified structure
*
* Input:
* - XXX_inst : Pointer to created instance that should be created
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_CreateEnc(CNG_enc_inst** cng_inst);
int16_t WebRtcCng_CreateDec(CNG_dec_inst** cng_inst);
/****************************************************************************
* WebRtcCng_InitEnc/Dec(...)
*
* This function initializes a instance
*
* Input:
* - cng_inst : Instance that should be initialized
*
* - fs : 8000 for narrowband and 16000 for wideband
* - interval : generate SID data every interval ms
* - quality : Number of refl. coefs, maximum allowed is 12
*
* Output:
* - cng_inst : Initialized instance
*
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcCng_InitEnc(CNG_enc_inst* cng_inst, int fs, int16_t interval,
int16_t quality);
void WebRtcCng_InitDec(CNG_dec_inst* cng_inst);
/****************************************************************************
* WebRtcCng_FreeEnc/Dec(...)
*
* These functions frees the dynamic memory of a specified instance
*
* Input:
* - cng_inst : Pointer to created instance that should be freed
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_FreeEnc(CNG_enc_inst* cng_inst);
int16_t WebRtcCng_FreeDec(CNG_dec_inst* cng_inst);
/****************************************************************************
* WebRtcCng_Encode(...)
*
* These functions analyzes background noise
*
* Input:
* - cng_inst : Pointer to created instance
* - speech : Signal to be analyzed
* - nrOfSamples : Size of speech vector
* - forceSID : not zero to force SID frame and reset
*
* Output:
* - bytesOut : Nr of bytes to transmit, might be 0
*
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcCng_Encode(CNG_enc_inst* cng_inst, int16_t* speech,
size_t nrOfSamples, uint8_t* SIDdata,
size_t* bytesOut, int16_t forceSID);
/****************************************************************************
* WebRtcCng_UpdateSid(...)
*
* These functions updates the CN state, when a new SID packet arrives
*
* Input:
* - cng_inst : Pointer to created instance that should be freed
* - SID : SID packet, all headers removed
* - length : Length in bytes of SID packet
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_UpdateSid(CNG_dec_inst* cng_inst, uint8_t* SID,
size_t length);
/****************************************************************************
* WebRtcCng_Generate(...)
*
* These functions generates CN data when needed
*
* Input:
* - cng_inst : Pointer to created instance that should be freed
* - outData : pointer to area to write CN data
* - nrOfSamples : How much data to generate
* - new_period : >0 if a new period of CNG, will reset history
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcCng_Generate(CNG_dec_inst* cng_inst, int16_t* outData,
size_t nrOfSamples, int16_t new_period);
/*****************************************************************************
* WebRtcCng_GetErrorCodeEnc/Dec(...)
*
* This functions can be used to check the error code of a CNG instance. When
* a function returns -1 a error code will be set for that instance. The
* function below extract the code of the last error that occurred in the
* specified instance.
*
* Input:
* - CNG_inst : CNG enc/dec instance
*
* Return value : Error code
*/
int16_t WebRtcCng_GetErrorCodeEnc(CNG_enc_inst* cng_inst);
int16_t WebRtcCng_GetErrorCodeDec(CNG_dec_inst* cng_inst);
#ifdef __cplusplus
}
#endif
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INCLUDE_WEBRTC_CNG_H_

54
webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
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/*
* Copyright (c) 2015 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 MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/scoped_refptr.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
template <typename T>
class AudioDecoderIsacT final : public AudioDecoder {
public:
struct Config {
bool IsOk() const;
int sample_rate_hz = 16000;
};
explicit AudioDecoderIsacT(const Config& config);
virtual ~AudioDecoderIsacT() override;
bool HasDecodePlc() const override;
size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
void Reset() override;
int ErrorCode() override;
int SampleRateHz() const override;
size_t Channels() const override;
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override;
private:
typename T::instance_type* isac_state_;
int sample_rate_hz_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_

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webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
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/*
* Copyright (c) 2015 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 MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
#include "rtc_base/checks.h"
namespace webrtc {
template <typename T>
bool AudioDecoderIsacT<T>::Config::IsOk() const {
return (sample_rate_hz == 16000 || sample_rate_hz == 32000);
}
template <typename T>
AudioDecoderIsacT<T>::AudioDecoderIsacT(const Config& config)
: sample_rate_hz_(config.sample_rate_hz) {
RTC_CHECK(config.IsOk()) << "Unsupported sample rate "
<< config.sample_rate_hz;
RTC_CHECK_EQ(0, T::Create(&isac_state_));
T::DecoderInit(isac_state_);
RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz_));
}
template <typename T>
AudioDecoderIsacT<T>::~AudioDecoderIsacT() {
RTC_CHECK_EQ(0, T::Free(isac_state_));
}
template <typename T>
int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
int16_t temp_type = 1; // Default is speech.
int ret =
T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return ret;
}
template <typename T>
bool AudioDecoderIsacT<T>::HasDecodePlc() const {
return false;
}
template <typename T>
size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) {
return T::DecodePlc(isac_state_, decoded, num_frames);
}
template <typename T>
void AudioDecoderIsacT<T>::Reset() {
T::DecoderInit(isac_state_);
}
template <typename T>
int AudioDecoderIsacT<T>::ErrorCode() {
return T::GetErrorCode(isac_state_);
}
template <typename T>
int AudioDecoderIsacT<T>::SampleRateHz() const {
return sample_rate_hz_;
}
template <typename T>
size_t AudioDecoderIsacT<T>::Channels() const {
return 1;
}
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_

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webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.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 MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/scoped_refptr.h"
#include "api/units/time_delta.h"
#include "rtc_base/constructor_magic.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
template <typename T>
class AudioEncoderIsacT final : public AudioEncoder {
public:
// Allowed combinations of sample rate, frame size, and bit rate are
// - 16000 Hz, 30 ms, 10000-32000 bps
// - 16000 Hz, 60 ms, 10000-32000 bps
// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
struct Config {
bool IsOk() const;
int payload_type = 103;
int sample_rate_hz = 16000;
int frame_size_ms = 30;
int bit_rate = kDefaultBitRate; // Limit on the short-term average bit
// rate, in bits/s.
int max_payload_size_bytes = -1;
int max_bit_rate = -1;
};
explicit AudioEncoderIsacT(const Config& config);
~AudioEncoderIsacT() override;
int SampleRateHz() const override;
size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
void SetTargetBitrate(int target_bps) override;
void OnReceivedTargetAudioBitrate(int target_bps) override;
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
absl::optional<int64_t> bwe_period_ms) override;
void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
void Reset() override;
absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
private:
// This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
// STREAM_MAXW16_60MS for iSAC fix (60 ms).
static const size_t kSufficientEncodeBufferSizeBytes = 400;
static constexpr int kDefaultBitRate = 32000;
static constexpr int kMinBitrateBps = 10000;
static constexpr int MaxBitrateBps(int sample_rate_hz) {
return sample_rate_hz == 32000 ? 56000 : 32000;
}
void SetTargetBitrate(int target_bps, bool subtract_per_packet_overhead);
// Recreate the iSAC encoder instance with the given settings, and save them.
void RecreateEncoderInstance(const Config& config);
Config config_;
typename T::instance_type* isac_state_ = nullptr;
// Have we accepted input but not yet emitted it in a packet?
bool packet_in_progress_ = false;
// Timestamp of the first input of the currently in-progress packet.
uint32_t packet_timestamp_;
// Timestamp of the previously encoded packet.
uint32_t last_encoded_timestamp_;
// Cache the value of the "WebRTC-SendSideBwe-WithOverhead" field trial.
const bool send_side_bwe_with_overhead_ =
field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead");
// When we send a packet, expect this many bytes of headers to be added to it.
// Start out with a reasonable default that we can use until we receive a real
// value.
DataSize overhead_per_packet_ = DataSize::Bytes(28);
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_

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webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.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 MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
template <typename T>
bool AudioEncoderIsacT<T>::Config::IsOk() const {
if (max_bit_rate < 32000 && max_bit_rate != -1)
return false;
if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
return false;
switch (sample_rate_hz) {
case 16000:
if (max_bit_rate > 53400)
return false;
if (max_payload_size_bytes > 400)
return false;
return (frame_size_ms == 30 || frame_size_ms == 60) &&
(bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 32000));
case 32000:
if (max_bit_rate > 160000)
return false;
if (max_payload_size_bytes > 600)
return false;
return T::has_swb &&
(frame_size_ms == 30 &&
(bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 56000)));
default:
return false;
}
}
template <typename T>
AudioEncoderIsacT<T>::AudioEncoderIsacT(const Config& config) {
RecreateEncoderInstance(config);
}
template <typename T>
AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
RTC_CHECK_EQ(0, T::Free(isac_state_));
}
template <typename T>
int AudioEncoderIsacT<T>::SampleRateHz() const {
return T::EncSampRate(isac_state_);
}
template <typename T>
size_t AudioEncoderIsacT<T>::NumChannels() const {
return 1;
}
template <typename T>
size_t AudioEncoderIsacT<T>::Num10MsFramesInNextPacket() const {
const int samples_in_next_packet = T::GetNewFrameLen(isac_state_);
return static_cast<size_t>(rtc::CheckedDivExact(
samples_in_next_packet, rtc::CheckedDivExact(SampleRateHz(), 100)));
}
template <typename T>
size_t AudioEncoderIsacT<T>::Max10MsFramesInAPacket() const {
return 6; // iSAC puts at most 60 ms in a packet.
}
template <typename T>
int AudioEncoderIsacT<T>::GetTargetBitrate() const {
return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate;
}
template <typename T>
void AudioEncoderIsacT<T>::SetTargetBitrate(int target_bps) {
// Set target bitrate directly without subtracting per-packet overhead,
// because that's what AudioEncoderOpus does.
SetTargetBitrate(target_bps,
/*subtract_per_packet_overhead=*/false);
}
template <typename T>
void AudioEncoderIsacT<T>::OnReceivedTargetAudioBitrate(int target_bps) {
// Set target bitrate directly without subtracting per-packet overhead,
// because that's what AudioEncoderOpus does.
SetTargetBitrate(target_bps,
/*subtract_per_packet_overhead=*/false);
}
template <typename T>
void AudioEncoderIsacT<T>::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
absl::optional<int64_t> /*bwe_period_ms*/) {
// Set target bitrate, subtracting the per-packet overhead if
// WebRTC-SendSideBwe-WithOverhead is enabled, because that's what
// AudioEncoderOpus does.
SetTargetBitrate(
target_audio_bitrate_bps,
/*subtract_per_packet_overhead=*/send_side_bwe_with_overhead_);
}
template <typename T>
void AudioEncoderIsacT<T>::OnReceivedUplinkAllocation(
BitrateAllocationUpdate update) {
// Set target bitrate, subtracting the per-packet overhead if
// WebRTC-SendSideBwe-WithOverhead is enabled, because that's what
// AudioEncoderOpus does.
SetTargetBitrate(
update.target_bitrate.bps<int>(),
/*subtract_per_packet_overhead=*/send_side_bwe_with_overhead_);
}
template <typename T>
void AudioEncoderIsacT<T>::OnReceivedOverhead(
size_t overhead_bytes_per_packet) {
overhead_per_packet_ = DataSize::Bytes(overhead_bytes_per_packet);
}
template <typename T>
AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeImpl(
uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) {
if (!packet_in_progress_) {
// Starting a new packet; remember the timestamp for later.
packet_in_progress_ = true;
packet_timestamp_ = rtp_timestamp;
}
size_t encoded_bytes = encoded->AppendData(
kSufficientEncodeBufferSizeBytes, [&](rtc::ArrayView<uint8_t> encoded) {
int r = T::Encode(isac_state_, audio.data(), encoded.data());
RTC_CHECK_GE(r, 0) << "Encode failed (error code "
<< T::GetErrorCode(isac_state_) << ")";
return static_cast<size_t>(r);
});
if (encoded_bytes == 0)
return EncodedInfo();
// Got enough input to produce a packet. Return the saved timestamp from
// the first chunk of input that went into the packet.
packet_in_progress_ = false;
EncodedInfo info;
info.encoded_bytes = encoded_bytes;
info.encoded_timestamp = packet_timestamp_;
info.payload_type = config_.payload_type;
info.encoder_type = CodecType::kIsac;
return info;
}
template <typename T>
void AudioEncoderIsacT<T>::Reset() {
RecreateEncoderInstance(config_);
}
template <typename T>
absl::optional<std::pair<TimeDelta, TimeDelta>>
AudioEncoderIsacT<T>::GetFrameLengthRange() const {
return {{TimeDelta::Millis(config_.frame_size_ms),
TimeDelta::Millis(config_.frame_size_ms)}};
}
template <typename T>
void AudioEncoderIsacT<T>::SetTargetBitrate(int target_bps,
bool subtract_per_packet_overhead) {
if (subtract_per_packet_overhead) {
const DataRate overhead_rate =
overhead_per_packet_ / TimeDelta::Millis(config_.frame_size_ms);
target_bps -= overhead_rate.bps();
}
target_bps = rtc::SafeClamp(target_bps, kMinBitrateBps,
MaxBitrateBps(config_.sample_rate_hz));
int result = T::Control(isac_state_, target_bps, config_.frame_size_ms);
RTC_DCHECK_EQ(result, 0);
config_.bit_rate = target_bps;
}
template <typename T>
void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
RTC_CHECK(config.IsOk());
packet_in_progress_ = false;
if (isac_state_)
RTC_CHECK_EQ(0, T::Free(isac_state_));
RTC_CHECK_EQ(0, T::Create(&isac_state_));
RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, /*coding_mode=*/1));
RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
if (config.max_payload_size_bytes != -1)
RTC_CHECK_EQ(
0, T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
if (config.max_bit_rate != -1)
RTC_CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
// Set the decoder sample rate even though we just use the encoder. This
// doesn't appear to be necessary to produce a valid encoding, but without it
// we get an encoding that isn't bit-for-bit identical with what a combined
// encoder+decoder object produces.
RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
config_ = config;
}
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_

33
webrtc/rtc_base/ignore_wundef.h
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@ -1,33 +0,0 @@
/*
* Copyright (c) 2016 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 RTC_BASE_IGNORE_WUNDEF_H_
#define RTC_BASE_IGNORE_WUNDEF_H_
// If a header file uses #if on possibly undefined macros (and it's for some
// reason not possible to just fix the header file), include it like this:
//
// RTC_PUSH_IGNORING_WUNDEF()
// #include "misbehaving_header.h"
// RTC_POP_IGNORING_WUNDEF()
//
// This will cause the compiler to not emit -Wundef warnings for that file.
#ifdef __clang__
#define RTC_PUSH_IGNORING_WUNDEF() \
_Pragma("clang diagnostic push") \
_Pragma("clang diagnostic ignored \"-Wundef\"")
#define RTC_POP_IGNORING_WUNDEF() _Pragma("clang diagnostic pop")
#else
#define RTC_PUSH_IGNORING_WUNDEF()
#define RTC_POP_IGNORING_WUNDEF()
#endif // __clang__
#endif // RTC_BASE_IGNORE_WUNDEF_H_

1
webrtc/rtc_base/meson.build
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@ -15,7 +15,6 @@ base_sources = [
'string_utils.cc',
'strings/string_builder.cc',
'synchronization/sequence_checker_internal.cc',
'synchronization/yield.cc',
'synchronization/yield_policy.cc',
'system/file_wrapper.cc',
'system_time.cc',

36
webrtc/rtc_base/synchronization/yield.cc
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@ -1,36 +0,0 @@
/*
* Copyright 2020 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 "rtc_base/synchronization/yield.h"
#if defined(WEBRTC_WIN)
#include <windows.h>
#else
#include <sched.h>
#include <time.h>
#endif
namespace webrtc {
void YieldCurrentThread() {
// TODO(bugs.webrtc.org/11634): use dedicated OS functionality instead of
// sleep for yielding.
#if defined(WEBRTC_WIN)
::Sleep(0);
#elif defined(WEBRTC_MAC) && defined(RTC_USE_NATIVE_MUTEX_ON_MAC) && \
!RTC_USE_NATIVE_MUTEX_ON_MAC
sched_yield();
#else
static const struct timespec ts_null = {0};
nanosleep(&ts_null, nullptr);
#endif
}
} // namespace webrtc

20
webrtc/rtc_base/synchronization/yield.h
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@ -1,20 +0,0 @@
/*
* Copyright 2020 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 RTC_BASE_SYNCHRONIZATION_YIELD_H_
#define RTC_BASE_SYNCHRONIZATION_YIELD_H_
namespace webrtc {
// Request rescheduling of threads.
void YieldCurrentThread();
} // namespace webrtc
#endif // RTC_BASE_SYNCHRONIZATION_YIELD_H_

28
webrtc/rtc_base/system/unused.h
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@ -1,28 +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 RTC_BASE_SYSTEM_UNUSED_H_
#define RTC_BASE_SYSTEM_UNUSED_H_
// Prevent the compiler from warning about an unused variable. For example:
// int result = DoSomething();
// RTC_DCHECK(result == 17);
// RTC_UNUSED(result);
// Note: In most cases it is better to remove the unused variable rather than
// suppressing the compiler warning.
#ifndef RTC_UNUSED
#ifdef __cplusplus
#define RTC_UNUSED(x) static_cast<void>(x)
#else
#define RTC_UNUSED(x) (void)(x)
#endif
#endif // RTC_UNUSED
#endif // RTC_BASE_SYSTEM_UNUSED_H_