diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc')
| -rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc | 824 |
1 files changed, 824 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc new file mode 100644 index 0000000000..17e0e33b1d --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc @@ -0,0 +1,824 @@ +/* + * 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) { + 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(); + + bool dtx_frame = (info.encoded_bytes <= 2); + + // Will use new packet size for next encoding. + config_.frame_size_ms = next_frame_length_ms_; + + 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 = !dtx_frame && (consecutive_dtx_frames_ != 20); + info.encoder_type = CodecType::kOpus; + + // Increase or reset DTX counter. + consecutive_dtx_frames_ = (dtx_frame) ? (consecutive_dtx_frames_ + 1) : (0); + + return info; +} + +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)}}; + } +} + +} // namespace webrtc |