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webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
Jim Gustafson 7da0a87124
Add more audio control and safe defaults
2023-08-23 10:42:30 -07:00

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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 "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
#include <algorithm>
#include <iterator>
#include <memory>
#include <string>
#include <utility>
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/exp_filter.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/string_to_number.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
// Codec parameters for Opus.
// draft-spittka-payload-rtp-opus-03
// Recommended bitrates:
// 8-12 kb/s for NB speech,
// 16-20 kb/s for WB speech,
// 28-40 kb/s for FB speech,
// 48-64 kb/s for FB mono music, and
// 64-128 kb/s for FB stereo music.
// The current implementation applies the following values to mono signals,
// and multiplies them by 2 for stereo.
constexpr int kOpusBitrateNbBps = 12000;
constexpr int kOpusBitrateWbBps = 20000;
constexpr int kOpusBitrateFbBps = 32000;
constexpr int kRtpTimestampRateHz = 48000;
constexpr int kDefaultMaxPlaybackRate = 48000;
// These two lists must be sorted from low to high
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
constexpr int kANASupportedFrameLengths[] = {20, 40, 60, 120};
constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120};
#else
constexpr int kANASupportedFrameLengths[] = {20, 40, 60};
constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
#endif
// PacketLossFractionSmoother uses an exponential filter with a time constant
// of -1.0 / ln(0.9999) = 10000 ms.
constexpr float kAlphaForPacketLossFractionSmoother = 0.9999f;
constexpr float kMaxPacketLossFraction = 0.2f;
int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
const int bitrate = [&] {
if (max_playback_rate <= 8000) {
return kOpusBitrateNbBps * rtc::dchecked_cast<int>(num_channels);
} else if (max_playback_rate <= 16000) {
return kOpusBitrateWbBps * rtc::dchecked_cast<int>(num_channels);
} else {
return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
}
}();
RTC_DCHECK_GE(bitrate, AudioEncoderOpusConfig::kMinBitrateBps);
RTC_DCHECK_LE(bitrate, AudioEncoderOpusConfig::kMaxBitrateBps);
return bitrate;
}
// Get the maxaveragebitrate parameter in string-form, so we can properly figure
// out how invalid it is and accurately log invalid values.
int CalculateBitrate(int max_playback_rate_hz,
size_t num_channels,
absl::optional<std::string> bitrate_param) {
const int default_bitrate =
CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
if (bitrate_param) {
const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
if (bitrate) {
const int chosen_bitrate =
std::max(AudioEncoderOpusConfig::kMinBitrateBps,
std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
if (bitrate != chosen_bitrate) {
RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
<< " clamped to " << chosen_bitrate;
}
return chosen_bitrate;
}
RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate \"" << *bitrate_param
<< "\" replaced by default bitrate " << default_bitrate;
}
return default_bitrate;
}
int GetChannelCount(const SdpAudioFormat& format) {
const auto param = GetFormatParameter(format, "stereo");
if (param == "1") {
return 2;
} else {
return 1;
}
}
int GetMaxPlaybackRate(const SdpAudioFormat& format) {
const auto param = GetFormatParameter<int>(format, "maxplaybackrate");
if (param && *param >= 8000) {
return std::min(*param, kDefaultMaxPlaybackRate);
}
return kDefaultMaxPlaybackRate;
}
int GetFrameSizeMs(const SdpAudioFormat& format) {
const auto ptime = GetFormatParameter<int>(format, "ptime");
if (ptime) {
// Pick the next highest supported frame length from
// kOpusSupportedFrameLengths.
for (const int supported_frame_length : kOpusSupportedFrameLengths) {
if (supported_frame_length >= *ptime) {
return supported_frame_length;
}
}
// If none was found, return the largest supported frame length.
return *(std::end(kOpusSupportedFrameLengths) - 1);
}
return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
}
void FindSupportedFrameLengths(int min_frame_length_ms,
int max_frame_length_ms,
std::vector<int>* out) {
out->clear();
std::copy_if(std::begin(kANASupportedFrameLengths),
std::end(kANASupportedFrameLengths), std::back_inserter(*out),
[&](int frame_length_ms) {
return frame_length_ms >= min_frame_length_ms &&
frame_length_ms <= max_frame_length_ms;
});
RTC_DCHECK(std::is_sorted(out->begin(), out->end()));
}
int GetBitrateBps(const AudioEncoderOpusConfig& config) {
RTC_DCHECK(config.IsOk());
return *config.bitrate_bps;
}
std::vector<float> GetBitrateMultipliers() {
constexpr char kBitrateMultipliersName[] =
"WebRTC-Audio-OpusBitrateMultipliers";
const bool use_bitrate_multipliers =
webrtc::field_trial::IsEnabled(kBitrateMultipliersName);
if (use_bitrate_multipliers) {
const std::string field_trial_string =
webrtc::field_trial::FindFullName(kBitrateMultipliersName);
std::vector<std::string> pieces;
rtc::tokenize(field_trial_string, '-', &pieces);
if (pieces.size() < 2 || pieces[0] != "Enabled") {
RTC_LOG(LS_WARNING) << "Invalid parameters for "
<< kBitrateMultipliersName
<< ", not using custom values.";
return std::vector<float>();
}
std::vector<float> multipliers(pieces.size() - 1);
for (size_t i = 1; i < pieces.size(); i++) {
if (!rtc::FromString(pieces[i], &multipliers[i - 1])) {
RTC_LOG(LS_WARNING)
<< "Invalid parameters for " << kBitrateMultipliersName
<< ", not using custom values.";
return std::vector<float>();
}
}
RTC_LOG(LS_INFO) << "Using custom bitrate multipliers: "
<< field_trial_string;
return multipliers;
}
return std::vector<float>();
}
int GetMultipliedBitrate(int bitrate, const std::vector<float>& multipliers) {
// The multipliers are valid from 5 kbps.
const size_t bitrate_kbps = static_cast<size_t>(bitrate / 1000);
if (bitrate_kbps < 5 || bitrate_kbps >= multipliers.size() + 5) {
return bitrate;
}
return static_cast<int>(multipliers[bitrate_kbps - 5] * bitrate);
}
} // namespace
void AudioEncoderOpusImpl::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
const SdpAudioFormat fmt = {"opus",
kRtpTimestampRateHz,
2,
{{"minptime", "10"}, {"useinbandfec", "1"}}};
const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
specs->push_back({fmt, info});
}
AudioCodecInfo AudioEncoderOpusImpl::QueryAudioEncoder(
const AudioEncoderOpusConfig& config) {
RTC_DCHECK(config.IsOk());
AudioCodecInfo info(config.sample_rate_hz, config.num_channels,
*config.bitrate_bps,
AudioEncoderOpusConfig::kMinBitrateBps,
AudioEncoderOpusConfig::kMaxBitrateBps);
info.allow_comfort_noise = false;
info.supports_network_adaption = true;
return info;
}
std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
const AudioEncoderOpusConfig& config,
int payload_type) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;
}
return std::make_unique<AudioEncoderOpusImpl>(config, payload_type);
}
absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "opus") ||
format.clockrate_hz != kRtpTimestampRateHz || format.num_channels != 2) {
return absl::nullopt;
}
AudioEncoderOpusConfig config;
config.num_channels = GetChannelCount(format);
config.frame_size_ms = GetFrameSizeMs(format);
config.max_playback_rate_hz = GetMaxPlaybackRate(format);
config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
config.bitrate_bps =
CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
GetFormatParameter(format, "maxaveragebitrate"));
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
constexpr int kMinANAFrameLength = kANASupportedFrameLengths[0];
constexpr int kMaxANAFrameLength =
kANASupportedFrameLengths[arraysize(kANASupportedFrameLengths) - 1];
// For now, minptime and maxptime are only used with ANA. If ptime is outside
// of this range, it will get adjusted once ANA takes hold. Ideally, we'd know
// if ANA was to be used when setting up the config, and adjust accordingly.
const int min_frame_length_ms =
GetFormatParameter<int>(format, "minptime").value_or(kMinANAFrameLength);
const int max_frame_length_ms =
GetFormatParameter<int>(format, "maxptime").value_or(kMaxANAFrameLength);
FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
&config.supported_frame_lengths_ms);
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return absl::nullopt;
}
return config;
}
absl::optional<int> AudioEncoderOpusImpl::GetNewComplexity(
const AudioEncoderOpusConfig& config) {
RTC_DCHECK(config.IsOk());
const int bitrate_bps = GetBitrateBps(config);
if (bitrate_bps >= config.complexity_threshold_bps -
config.complexity_threshold_window_bps &&
bitrate_bps <= config.complexity_threshold_bps +
config.complexity_threshold_window_bps) {
// Within the hysteresis window; make no change.
return absl::nullopt;
} else {
return bitrate_bps <= config.complexity_threshold_bps
? config.low_rate_complexity
: config.complexity;
}
}
absl::optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
const AudioEncoderOpusConfig& config,
OpusEncInst* inst) {
constexpr int kMinWidebandBitrate = 8000;
constexpr int kMaxNarrowbandBitrate = 9000;
constexpr int kAutomaticThreshold = 11000;
RTC_DCHECK(config.IsOk());
const int bitrate = GetBitrateBps(config);
if (bitrate > kAutomaticThreshold) {
return absl::optional<int>(OPUS_AUTO);
}
const int bandwidth = WebRtcOpus_GetBandwidth(inst);
RTC_DCHECK_GE(bandwidth, 0);
if (bitrate > kMaxNarrowbandBitrate && bandwidth < OPUS_BANDWIDTH_WIDEBAND) {
return absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND);
} else if (bitrate < kMinWidebandBitrate &&
bandwidth > OPUS_BANDWIDTH_NARROWBAND) {
return absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND);
}
return absl::optional<int>();
}
class AudioEncoderOpusImpl::PacketLossFractionSmoother {
public:
explicit PacketLossFractionSmoother()
: last_sample_time_ms_(rtc::TimeMillis()),
smoother_(kAlphaForPacketLossFractionSmoother) {}
// Gets the smoothed packet loss fraction.
float GetAverage() const {
float value = smoother_.filtered();
return (value == rtc::ExpFilter::kValueUndefined) ? 0.0f : value;
}
// Add new observation to the packet loss fraction smoother.
void AddSample(float packet_loss_fraction) {
int64_t now_ms = rtc::TimeMillis();
smoother_.Apply(static_cast<float>(now_ms - last_sample_time_ms_),
packet_loss_fraction);
last_sample_time_ms_ = now_ms;
}
private:
int64_t last_sample_time_ms_;
// An exponential filter is used to smooth the packet loss fraction.
rtc::ExpFilter smoother_;
};
AudioEncoderOpusImpl::AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config,
int payload_type)
: AudioEncoderOpusImpl(
config,
payload_type,
[this](absl::string_view config_string, RtcEventLog* event_log) {
return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
},
// We choose 5sec as initial time constant due to empirical data.
std::make_unique<SmoothingFilterImpl>(5000)) {}
AudioEncoderOpusImpl::AudioEncoderOpusImpl(
const AudioEncoderOpusConfig& config,
int payload_type,
const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
std::unique_ptr<SmoothingFilter> bitrate_smoother)
: payload_type_(payload_type),
use_stable_target_for_adaptation_(!webrtc::field_trial::IsDisabled(
"WebRTC-Audio-StableTargetAdaptation")),
adjust_bandwidth_(
webrtc::field_trial::IsEnabled("WebRTC-AdjustOpusBandwidth")),
bitrate_changed_(true),
bitrate_multipliers_(GetBitrateMultipliers()),
packet_loss_rate_(0.0),
inst_(nullptr),
packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
audio_network_adaptor_creator_(audio_network_adaptor_creator),
bitrate_smoother_(std::move(bitrate_smoother)),
consecutive_dtx_frames_(0) {
RTC_DCHECK(0 <= payload_type && payload_type <= 127);
// Sanity check of the redundant payload type field that we want to get rid
// of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
RTC_CHECK(RecreateEncoderInstance(config));
SetProjectedPacketLossRate(packet_loss_rate_);
}
AudioEncoderOpusImpl::AudioEncoderOpusImpl(int payload_type,
const SdpAudioFormat& format)
: AudioEncoderOpusImpl(*SdpToConfig(format), payload_type) {}
AudioEncoderOpusImpl::~AudioEncoderOpusImpl() {
RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
}
int AudioEncoderOpusImpl::SampleRateHz() const {
return config_.sample_rate_hz;
}
size_t AudioEncoderOpusImpl::NumChannels() const {
return config_.num_channels;
}
int AudioEncoderOpusImpl::RtpTimestampRateHz() const {
return kRtpTimestampRateHz;
}
size_t AudioEncoderOpusImpl::Num10MsFramesInNextPacket() const {
return Num10msFramesPerPacket();
}
size_t AudioEncoderOpusImpl::Max10MsFramesInAPacket() const {
return Num10msFramesPerPacket();
}
int AudioEncoderOpusImpl::GetTargetBitrate() const {
return GetBitrateBps(config_);
}
void AudioEncoderOpusImpl::Reset() {
RTC_CHECK(RecreateEncoderInstance(config_));
}
bool AudioEncoderOpusImpl::SetFec(bool enable) {
if (enable) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
}
config_.fec_enabled = enable;
return true;
}
bool AudioEncoderOpusImpl::SetDtx(bool enable) {
if (enable) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
}
config_.dtx_enabled = enable;
return true;
}
bool AudioEncoderOpusImpl::GetDtx() const {
return config_.dtx_enabled;
}
bool AudioEncoderOpusImpl::SetApplication(Application application) {
auto conf = config_;
switch (application) {
case Application::kSpeech:
conf.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
break;
case Application::kAudio:
conf.application = AudioEncoderOpusConfig::ApplicationMode::kAudio;
break;
}
return RecreateEncoderInstance(conf);
}
void AudioEncoderOpusImpl::SetMaxPlaybackRate(int frequency_hz) {
auto conf = config_;
conf.max_playback_rate_hz = frequency_hz;
RTC_CHECK(RecreateEncoderInstance(conf));
}
bool AudioEncoderOpusImpl::EnableAudioNetworkAdaptor(
const std::string& config_string,
RtcEventLog* event_log) {
audio_network_adaptor_ =
audio_network_adaptor_creator_(config_string, event_log);
return audio_network_adaptor_.get() != nullptr;
}
void AudioEncoderOpusImpl::DisableAudioNetworkAdaptor() {
audio_network_adaptor_.reset(nullptr);
}
void AudioEncoderOpusImpl::OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) {
if (audio_network_adaptor_) {
audio_network_adaptor_->SetUplinkPacketLossFraction(
uplink_packet_loss_fraction);
ApplyAudioNetworkAdaptor();
}
packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction);
float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage();
SetProjectedPacketLossRate(average_fraction_loss);
}
void AudioEncoderOpusImpl::OnReceivedTargetAudioBitrate(
int target_audio_bitrate_bps) {
SetTargetBitrate(target_audio_bitrate_bps);
}
void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
absl::optional<int64_t> bwe_period_ms,
absl::optional<int64_t> stable_target_bitrate_bps) {
if (audio_network_adaptor_) {
audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
if (use_stable_target_for_adaptation_) {
if (stable_target_bitrate_bps)
audio_network_adaptor_->SetUplinkBandwidth(*stable_target_bitrate_bps);
} else {
// We give smoothed bitrate allocation to audio network adaptor as
// the uplink bandwidth.
// The BWE spikes should not affect the bitrate smoother more than 25%.
// To simplify the calculations we use a step response as input signal.
// The step response of an exponential filter is
// u(t) = 1 - e^(-t / time_constant).
// In order to limit the affect of a BWE spike within 25% of its value
// before
// the next BWE update, we would choose a time constant that fulfills
// 1 - e^(-bwe_period_ms / time_constant) < 0.25
// Then 4 * bwe_period_ms is a good choice.
if (bwe_period_ms)
bitrate_smoother_->SetTimeConstantMs(*bwe_period_ms * 4);
bitrate_smoother_->AddSample(target_audio_bitrate_bps);
}
ApplyAudioNetworkAdaptor();
} else {
if (!overhead_bytes_per_packet_) {
RTC_LOG(LS_INFO)
<< "AudioEncoderOpusImpl: Overhead unknown, target audio bitrate "
<< target_audio_bitrate_bps << " bps is ignored.";
return;
}
const int overhead_bps = static_cast<int>(
*overhead_bytes_per_packet_ * 8 * 100 / Num10MsFramesInNextPacket());
SetTargetBitrate(
std::min(AudioEncoderOpusConfig::kMaxBitrateBps,
std::max(AudioEncoderOpusConfig::kMinBitrateBps,
target_audio_bitrate_bps - overhead_bps)));
}
}
void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
absl::optional<int64_t> bwe_period_ms) {
OnReceivedUplinkBandwidth(target_audio_bitrate_bps, bwe_period_ms,
absl::nullopt);
}
void AudioEncoderOpusImpl::OnReceivedUplinkAllocation(
BitrateAllocationUpdate update) {
OnReceivedUplinkBandwidth(update.target_bitrate.bps(), update.bwe_period.ms(),
update.stable_target_bitrate.bps());
}
void AudioEncoderOpusImpl::OnReceivedRtt(int rtt_ms) {
if (!audio_network_adaptor_)
return;
audio_network_adaptor_->SetRtt(rtt_ms);
ApplyAudioNetworkAdaptor();
}
void AudioEncoderOpusImpl::OnReceivedOverhead(
size_t overhead_bytes_per_packet) {
if (audio_network_adaptor_) {
audio_network_adaptor_->SetOverhead(overhead_bytes_per_packet);
ApplyAudioNetworkAdaptor();
} else {
overhead_bytes_per_packet_ = overhead_bytes_per_packet;
}
}
void AudioEncoderOpusImpl::SetReceiverFrameLengthRange(
int min_frame_length_ms,
int max_frame_length_ms) {
// Ensure that `SetReceiverFrameLengthRange` is called before
// `EnableAudioNetworkAdaptor`, otherwise we need to recreate
// `audio_network_adaptor_`, which is not a needed use case.
RTC_DCHECK(!audio_network_adaptor_);
FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
&config_.supported_frame_lengths_ms);
}
AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) {
MaybeUpdateUplinkBandwidth();
if (input_buffer_.empty())
first_timestamp_in_buffer_ = rtp_timestamp;
input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
if (input_buffer_.size() <
(Num10msFramesPerPacket() * SamplesPer10msFrame())) {
return EncodedInfo();
}
RTC_CHECK_EQ(input_buffer_.size(),
Num10msFramesPerPacket() * SamplesPer10msFrame());
const size_t max_encoded_bytes = SufficientOutputBufferSize();
EncodedInfo info;
info.encoded_bytes = encoded->AppendData(
max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
int status = WebRtcOpus_Encode(
inst_, &input_buffer_[0],
rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
RTC_CHECK_GE(status, 0); // Fails only if fed invalid data.
return static_cast<size_t>(status);
});
input_buffer_.clear();
if (adjust_bandwidth_ && bitrate_changed_) {
const auto bandwidth = GetNewBandwidth(config_, inst_);
if (bandwidth) {
RTC_CHECK_EQ(0, WebRtcOpus_SetBandwidth(inst_, *bandwidth));
}
bitrate_changed_ = false;
}
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
info.send_even_if_empty = true; // Allows Opus to send empty packets.
// After 20 DTX frames (MAX_CONSECUTIVE_DTX) Opus will send a frame
// coding the background noise. Avoid flagging this frame as speech
// (even though there is a probability of the frame being speech).
info.speech = IsPacketSpeech(info.encoded_bytes, encoded->data());
info.encoder_type = CodecType::kOpus;
// Will use new packet size for next encoding.
config_.frame_size_ms = next_frame_length_ms_;
return info;
}
// RingRTC change to detect if an encoded packet contains speech or not.
// Generally, if the last frame in the packet is audio, it is speech, unless
// it is a DTX refresh frame. This function follows RFC-6716 to check frames
// in each encoded packet.
bool AudioEncoderOpusImpl::IsPacketSpeech(
int encoded_bytes,
const uint8_t* encoded) {
bool speech = false;
// If the encoder returns 0, 1, or 2 encoded bytes, by definition, the packet
// contains only DTX frame(s). 0 is the special case in which
// opus_interface.cc detected consecutive DTX packets and is instructing
// WebRTC not to send any packet out over the wire.
bool dtx_packet = (encoded_bytes <= 2);
if (config_.frame_size_ms > 20) {
// For packet times greater than 20ms, Opus will encode a group of 20ms
// frames and combine them into a 'packet' with a TOC.
if (dtx_packet) {
// The 'packet' contains only DTX frames.
consecutive_dtx_frames_ += config_.frame_size_ms / 20;
} else {
// The 'packet' contains at least one non-DTX frame.
if (((encoded[0] & 0x98) == 0x08) || // config values of 1, 5, 9, and 13
((encoded[0] & 0x78) == 0x78) || // config value of 15
((encoded[0] & 0x98) == 0x98)) { // config values of 19, 23, 27, and 31
// The TOC indicates a packet with 20ms frames.
int code = encoded[0] & 0x03;
if (code == 0) {
// Code 0: 1 frame in the packet
// This case is unlikely for DTX.
consecutive_dtx_frames_ = 0;
speech = true;
} else if (code == 1) {
// Code 1: 2 frames in the packet, each with equal compressed size
// If both frames were DTX, we would not reach here.
consecutive_dtx_frames_ = 0;
speech = true;
} else if (code == 2) {
// Code 2: 2 frames in the packet, with different compressed sizes
int header_bytes = 2;
int size_of_first_frame = encoded[1];
if (size_of_first_frame > 251) {
size_of_first_frame += encoded[2] * 4;
header_bytes = 3;
}
int size_of_second_frame = encoded_bytes - size_of_first_frame - header_bytes;
if (size_of_first_frame > 0 && size_of_second_frame > 0) {
// The second frame has to be speech.
consecutive_dtx_frames_ = 0;
speech = true;
} else if (size_of_first_frame == 0 && size_of_second_frame > 0) {
// Second frame may or may not be DTX refresh.
speech = (consecutive_dtx_frames_ + 1) != 20;
consecutive_dtx_frames_ = 0;
} else if (size_of_first_frame > 0 && size_of_second_frame == 0) {
// First frame may or may not be DTX refresh.
consecutive_dtx_frames_ = 1;
} else {
// Both frames are size 0/DTX, should not reach here.
consecutive_dtx_frames_ += 2;
}
} else if (code == 3) {
// Code 3: an arbitrary number of frames in the packet
bool variable = (encoded[1] & 0x80) == 0x80;
bool padding = (encoded[1] & 0x40) == 0x40;
int M = encoded[1] & 0x3f;
int padding_header_bytes = 0;
int padding_size = 0;
if (padding) {
if (encoded[2] == 0xff) {
if (encoded_bytes < 4) {
// The packet should be at least 4 bytes, reset.
consecutive_dtx_frames_ = 0;
return true;
}
padding_size = 254 + encoded[3];
padding_header_bytes = 2;
} else {
padding_size = encoded[2];
padding_header_bytes = 1;
}
}
if (variable) {
// Frames in the packet have a variable size, a mix of audio and DTX.
int offset = 2 + padding_header_bytes;
int frame_header_bytes = 0;
int total_size_of_frames = 0;
// Check the worst-case limits to be sure there is enough encoded
// data to evaluate.
if (encoded_bytes < offset + M * 2) {
// Note: This assumes that actual encoded data is larger than
// the guess of two bytes for each header... Reset.
consecutive_dtx_frames_ = 0;
return true;
}
// The only time we walk the packet header to check for dynamic frame
// sizes. Only expected for packets with at least one DTX frame and
// at least one audio/refresh frame.
for (int frame = 0; frame < M - 1; frame++) {
int frame_size = encoded[offset];
if (frame_size > 251) {
frame_size += encoded[++offset] * 4;
frame_header_bytes += 2;
} else {
frame_header_bytes += 1;
}
if (frame_size > 0) {
// Could be speech or a DTX refresh frame. In either case,
// reset the DTX count.
consecutive_dtx_frames_ = 0;
} else {
// DTX frame.
consecutive_dtx_frames_++;
}
total_size_of_frames += frame_size;
offset++;
}
// Then, the last frame size should be:
int frame_M_size = encoded_bytes - 2
- (padding_header_bytes + padding_size)
- (frame_header_bytes + total_size_of_frames);
if (frame_M_size > 0) {
// The packet is ending, could be speech or a DTX refresh frame.
speech = consecutive_dtx_frames_ != 20;
consecutive_dtx_frames_ = 0;
} else if (frame_M_size == 0) {
// The packet is ending on a DTX frame.
consecutive_dtx_frames_++;
} else {
// Badly formatted packet, reset.
consecutive_dtx_frames_ = 0;
return true;
}
} else {
// Frames in the packet have a constant size.
int R = encoded_bytes - 2 - (padding_header_bytes + padding_size);
if (R > 0) {
// All frames are the same size and larger than zero, so they must
// represent speech.
consecutive_dtx_frames_ = 0;
speech = true;
} else {
// All frames are DTX.
consecutive_dtx_frames_ += M;
}
}
}
} else {
// The TOC indicates a packet with something other than 20ms frames.
// This does not match the supported frame sizing, reset and consider
// the packet to represent speech.
consecutive_dtx_frames_ = 0;
speech = true;
}
}
} else {
speech = !dtx_packet && (consecutive_dtx_frames_ != 20);
consecutive_dtx_frames_ = (dtx_packet) ? (consecutive_dtx_frames_ + 1) : (0);
}
return speech;
}
size_t AudioEncoderOpusImpl::Num10msFramesPerPacket() const {
return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
}
size_t AudioEncoderOpusImpl::SamplesPer10msFrame() const {
return rtc::CheckedDivExact(config_.sample_rate_hz, 100) *
config_.num_channels;
}
size_t AudioEncoderOpusImpl::SufficientOutputBufferSize() const {
// Calculate the number of bytes we expect the encoder to produce,
// then multiply by two to give a wide margin for error.
const size_t bytes_per_millisecond =
static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
const size_t approx_encoded_bytes =
Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
return 2 * approx_encoded_bytes;
}
// If the given config is OK, recreate the Opus encoder instance with those
// settings, save the config, and return true. Otherwise, do nothing and return
// false.
bool AudioEncoderOpusImpl::RecreateEncoderInstance(
const AudioEncoderOpusConfig& config) {
if (!config.IsOk())
return false;
config_ = config;
if (inst_)
RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
input_buffer_.clear();
input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(
&inst_, config.num_channels,
config.application ==
AudioEncoderOpusConfig::ApplicationMode::kVoip
? 0
: 1,
config.sample_rate_hz));
const int bitrate = GetBitrateBps(config);
RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, bitrate));
RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
if (config.fec_enabled) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
}
RTC_CHECK_EQ(
0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
// Use the default complexity if the start bitrate is within the hysteresis
// window.
complexity_ = GetNewComplexity(config).value_or(config.complexity);
RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
bitrate_changed_ = true;
if (config.dtx_enabled) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
}
RTC_CHECK_EQ(0,
WebRtcOpus_SetPacketLossRate(
inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
if (config.cbr_enabled) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_));
} else {
RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_));
}
num_channels_to_encode_ = NumChannels();
next_frame_length_ms_ = config_.frame_size_ms;
return true;
}
void AudioEncoderOpusImpl::SetFrameLength(int frame_length_ms) {
if (next_frame_length_ms_ != frame_length_ms) {
RTC_LOG(LS_VERBOSE) << "Update Opus frame length "
<< "from " << next_frame_length_ms_ << " ms "
<< "to " << frame_length_ms << " ms.";
}
next_frame_length_ms_ = frame_length_ms;
}
void AudioEncoderOpusImpl::SetNumChannelsToEncode(
size_t num_channels_to_encode) {
RTC_DCHECK_GT(num_channels_to_encode, 0);
RTC_DCHECK_LE(num_channels_to_encode, config_.num_channels);
if (num_channels_to_encode_ == num_channels_to_encode)
return;
RTC_CHECK_EQ(0, WebRtcOpus_SetForceChannels(inst_, num_channels_to_encode));
num_channels_to_encode_ = num_channels_to_encode;
}
void AudioEncoderOpusImpl::SetProjectedPacketLossRate(float fraction) {
fraction = std::min(std::max(fraction, 0.0f), kMaxPacketLossFraction);
if (packet_loss_rate_ != fraction) {
packet_loss_rate_ = fraction;
RTC_CHECK_EQ(
0, WebRtcOpus_SetPacketLossRate(
inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
}
}
void AudioEncoderOpusImpl::SetTargetBitrate(int bits_per_second) {
const int new_bitrate = rtc::SafeClamp<int>(
bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
AudioEncoderOpusConfig::kMaxBitrateBps);
if (config_.bitrate_bps && *config_.bitrate_bps != new_bitrate) {
config_.bitrate_bps = new_bitrate;
RTC_DCHECK(config_.IsOk());
const int bitrate = GetBitrateBps(config_);
RTC_CHECK_EQ(
0, WebRtcOpus_SetBitRate(
inst_, GetMultipliedBitrate(bitrate, bitrate_multipliers_)));
RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
bitrate_changed_ = true;
}
const auto new_complexity = GetNewComplexity(config_);
if (new_complexity && complexity_ != *new_complexity) {
complexity_ = *new_complexity;
RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
}
}
void AudioEncoderOpusImpl::ApplyAudioNetworkAdaptor() {
auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
if (config.bitrate_bps)
SetTargetBitrate(*config.bitrate_bps);
if (config.frame_length_ms)
SetFrameLength(*config.frame_length_ms);
if (config.enable_dtx)
SetDtx(*config.enable_dtx);
if (config.num_channels)
SetNumChannelsToEncode(*config.num_channels);
}
std::unique_ptr<AudioNetworkAdaptor>
AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator(
absl::string_view config_string,
RtcEventLog* event_log) const {
AudioNetworkAdaptorImpl::Config config;
config.event_log = event_log;
return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
config, ControllerManagerImpl::Create(
config_string, NumChannels(), supported_frame_lengths_ms(),
AudioEncoderOpusConfig::kMinBitrateBps,
num_channels_to_encode_, next_frame_length_ms_,
GetTargetBitrate(), config_.fec_enabled, GetDtx())));
}
void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
if (audio_network_adaptor_ && !use_stable_target_for_adaptation_) {
int64_t now_ms = rtc::TimeMillis();
if (!bitrate_smoother_last_update_time_ ||
now_ms - *bitrate_smoother_last_update_time_ >=
config_.uplink_bandwidth_update_interval_ms) {
absl::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
if (smoothed_bitrate)
audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
bitrate_smoother_last_update_time_ = now_ms;
}
}
}
ANAStats AudioEncoderOpusImpl::GetANAStats() const {
if (audio_network_adaptor_) {
return audio_network_adaptor_->GetStats();
}
return ANAStats();
}
absl::optional<std::pair<TimeDelta, TimeDelta> >
AudioEncoderOpusImpl::GetFrameLengthRange() const {
if (audio_network_adaptor_) {
if (config_.supported_frame_lengths_ms.empty()) {
return absl::nullopt;
}
return {{TimeDelta::Millis(config_.supported_frame_lengths_ms.front()),
TimeDelta::Millis(config_.supported_frame_lengths_ms.back())}};
} else {
return {{TimeDelta::Millis(config_.frame_size_ms),
TimeDelta::Millis(config_.frame_size_ms)}};
}
}
// RingRTC change to configure opus
bool AudioEncoderOpusImpl::Configure(const webrtc::AudioEncoder::Config& config) {
// This sets next_frame_length_ms_ until the next time audio is sampled,
// and then it sets config_.frame_size_ms as well.
// It needs to be delayed to avoid a CHECK in Encode.
SetFrameLength(config.initial_packet_size_ms);
// I don't think any of the below are necessary, but the above is, so we might as well set these.
config_.bitrate_bps = config.initial_bitrate_bps;
config_.fec_enabled = config.enable_fec;
config_.cbr_enabled = config.enable_cbr;
config_.complexity = config.complexity;
config_.low_rate_complexity = config_.low_rate_complexity;
config_.dtx_enabled = config.enable_dtx;
if (config.adaptation > 0) {
RTC_LOG(LS_WARNING) << "ringrtc_adapt!,audio,0," << config.initial_bitrate_bps
<< "," << config.initial_packet_size_ms;
}
if (WebRtcOpus_SetBandwidth(inst_, config.bandwidth) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to bandwidth=" << config.bandwidth;
return false;
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to bandwidth=" << config.bandwidth;
if (WebRtcOpus_SetBitRate(inst_, config.initial_bitrate_bps) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to bitrate_bps=" << config.initial_bitrate_bps;
return false;
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to bitrate_bps=" << config.initial_bitrate_bps;
if (WebRtcOpus_SetComplexity(inst_, config.complexity) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to complexity=" << config.complexity;
return false;
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to complexity=" << config.complexity;
if (config.enable_fec) {
if (WebRtcOpus_EnableFec(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_fec=" << config.enable_fec;
return false;
}
} else {
if (WebRtcOpus_DisableFec(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_fec=" << config.enable_fec;
return false;
}
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to enable_fec=" << config.enable_fec;
if (config.enable_dtx) {
if (WebRtcOpus_EnableDtx(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_dtx=" << config.enable_dtx;
return false;
}
} else {
if (WebRtcOpus_DisableDtx(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_dtx=" << config.enable_dtx;
return false;
}
}
RTC_LOG(LS_INFO) << "Successfully configured OPUS to enable_dtx=" << config.enable_dtx;
if (config.enable_cbr) {
if (WebRtcOpus_EnableCbr(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_cbr=" << config.enable_cbr;
return false;
}
} else {
if (WebRtcOpus_DisableCbr(inst_) == -1) {
RTC_LOG(LS_WARNING) << "Failed to configure OPUS to enable_cbr=" << config.enable_cbr;
return false;
}
}
return true;
}
} // namespace webrtc
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