| /* |
| * 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 "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h" |
| |
| #include <algorithm> |
| #include <iterator> |
| |
| #include "webrtc/base/analytics/exp_filter.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/safe_conversions.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h" |
| #include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h" |
| #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| constexpr int kSampleRateHz = 48000; |
| constexpr int kMinBitrateBps = 500; |
| constexpr int kMaxBitrateBps = 512000; |
| constexpr int kSupportedFrameLengths[] = {20, 60}; |
| |
| // PacketLossFractionSmoother uses an exponential filter with a time constant |
| // of -1.0 / ln(0.9999) = 10000 ms. |
| constexpr float kAlphaForPacketLossFractionSmoother = 0.9999f; |
| |
| AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) { |
| AudioEncoderOpus::Config config; |
| config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48); |
| config.num_channels = codec_inst.channels; |
| config.bitrate_bps = rtc::Optional<int>(codec_inst.rate); |
| config.payload_type = codec_inst.pltype; |
| config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip |
| : AudioEncoderOpus::kAudio; |
| config.supported_frame_lengths_ms.push_back(config.frame_size_ms); |
| #if WEBRTC_OPUS_VARIABLE_COMPLEXITY |
| config.low_rate_complexity = 9; |
| #endif |
| return config; |
| } |
| |
| // Optimize the loss rate to configure Opus. Basically, optimized loss rate is |
| // the input loss rate rounded down to various levels, because a robustly good |
| // audio quality is achieved by lowering the packet loss down. |
| // Additionally, to prevent toggling, margins are used, i.e., when jumping to |
| // a loss rate from below, a higher threshold is used than jumping to the same |
| // level from above. |
| double OptimizePacketLossRate(double new_loss_rate, double old_loss_rate) { |
| RTC_DCHECK_GE(new_loss_rate, 0.0); |
| RTC_DCHECK_LE(new_loss_rate, 1.0); |
| RTC_DCHECK_GE(old_loss_rate, 0.0); |
| RTC_DCHECK_LE(old_loss_rate, 1.0); |
| const double kPacketLossRate20 = 0.20; |
| const double kPacketLossRate10 = 0.10; |
| const double kPacketLossRate5 = 0.05; |
| const double kPacketLossRate1 = 0.01; |
| const double kLossRate20Margin = 0.02; |
| const double kLossRate10Margin = 0.01; |
| const double kLossRate5Margin = 0.01; |
| if (new_loss_rate >= |
| kPacketLossRate20 + |
| kLossRate20Margin * |
| (kPacketLossRate20 - old_loss_rate > 0 ? 1 : -1)) { |
| return kPacketLossRate20; |
| } else if (new_loss_rate >= |
| kPacketLossRate10 + |
| kLossRate10Margin * |
| (kPacketLossRate10 - old_loss_rate > 0 ? 1 : -1)) { |
| return kPacketLossRate10; |
| } else if (new_loss_rate >= |
| kPacketLossRate5 + |
| kLossRate5Margin * |
| (kPacketLossRate5 - old_loss_rate > 0 ? 1 : -1)) { |
| return kPacketLossRate5; |
| } else if (new_loss_rate >= kPacketLossRate1) { |
| return kPacketLossRate1; |
| } else { |
| return 0.0; |
| } |
| } |
| |
| } // namespace |
| |
| class AudioEncoderOpus::PacketLossFractionSmoother { |
| public: |
| explicit PacketLossFractionSmoother(const Clock* clock) |
| : clock_(clock), |
| last_sample_time_ms_(clock_->TimeInMilliseconds()), |
| 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 = clock_->TimeInMilliseconds(); |
| smoother_.Apply(static_cast<float>(now_ms - last_sample_time_ms_), |
| packet_loss_fraction); |
| last_sample_time_ms_ = now_ms; |
| } |
| |
| private: |
| const Clock* const clock_; |
| int64_t last_sample_time_ms_; |
| |
| // An exponential filter is used to smooth the packet loss fraction. |
| rtc::ExpFilter smoother_; |
| }; |
| |
| AudioEncoderOpus::Config::Config() { |
| #if WEBRTC_OPUS_VARIABLE_COMPLEXITY |
| low_rate_complexity = 9; |
| #endif |
| } |
| AudioEncoderOpus::Config::Config(const Config&) = default; |
| AudioEncoderOpus::Config::~Config() = default; |
| auto AudioEncoderOpus::Config::operator=(const Config&) -> Config& = default; |
| |
| bool AudioEncoderOpus::Config::IsOk() const { |
| if (frame_size_ms <= 0 || frame_size_ms % 10 != 0) |
| return false; |
| if (num_channels != 1 && num_channels != 2) |
| return false; |
| if (bitrate_bps && |
| (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps)) |
| return false; |
| if (complexity < 0 || complexity > 10) |
| return false; |
| if (low_rate_complexity < 0 || low_rate_complexity > 10) |
| return false; |
| return true; |
| } |
| |
| int AudioEncoderOpus::Config::GetBitrateBps() const { |
| RTC_DCHECK(IsOk()); |
| if (bitrate_bps) |
| return *bitrate_bps; // Explicitly set value. |
| else |
| return num_channels == 1 ? 32000 : 64000; // Default value. |
| } |
| |
| rtc::Optional<int> AudioEncoderOpus::Config::GetNewComplexity() const { |
| RTC_DCHECK(IsOk()); |
| const int bitrate_bps = GetBitrateBps(); |
| if (bitrate_bps >= |
| complexity_threshold_bps - complexity_threshold_window_bps && |
| bitrate_bps <= |
| complexity_threshold_bps + complexity_threshold_window_bps) { |
| // Within the hysteresis window; make no change. |
| return rtc::Optional<int>(); |
| } |
| return bitrate_bps <= complexity_threshold_bps |
| ? rtc::Optional<int>(low_rate_complexity) |
| : rtc::Optional<int>(complexity); |
| } |
| |
| AudioEncoderOpus::AudioEncoderOpus( |
| const Config& config, |
| AudioNetworkAdaptorCreator&& audio_network_adaptor_creator) |
| : packet_loss_rate_(0.0), |
| inst_(nullptr), |
| packet_loss_fraction_smoother_(new PacketLossFractionSmoother( |
| config.clock ? config.clock : Clock::GetRealTimeClock())), |
| audio_network_adaptor_creator_( |
| audio_network_adaptor_creator |
| ? std::move(audio_network_adaptor_creator) |
| : [this](const std::string& config_string, const Clock* clock) { |
| return DefaultAudioNetworkAdaptorCreator(config_string, |
| clock); |
| }) { |
| RTC_CHECK(RecreateEncoderInstance(config)); |
| } |
| |
| AudioEncoderOpus::AudioEncoderOpus(const CodecInst& codec_inst) |
| : AudioEncoderOpus(CreateConfig(codec_inst), nullptr) {} |
| |
| AudioEncoderOpus::~AudioEncoderOpus() { |
| RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_)); |
| } |
| |
| int AudioEncoderOpus::SampleRateHz() const { |
| return kSampleRateHz; |
| } |
| |
| size_t AudioEncoderOpus::NumChannels() const { |
| return config_.num_channels; |
| } |
| |
| size_t AudioEncoderOpus::Num10MsFramesInNextPacket() const { |
| return Num10msFramesPerPacket(); |
| } |
| |
| size_t AudioEncoderOpus::Max10MsFramesInAPacket() const { |
| return Num10msFramesPerPacket(); |
| } |
| |
| int AudioEncoderOpus::GetTargetBitrate() const { |
| return config_.GetBitrateBps(); |
| } |
| |
| void AudioEncoderOpus::Reset() { |
| RTC_CHECK(RecreateEncoderInstance(config_)); |
| } |
| |
| bool AudioEncoderOpus::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 AudioEncoderOpus::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 AudioEncoderOpus::GetDtx() const { |
| return config_.dtx_enabled; |
| } |
| |
| bool AudioEncoderOpus::SetApplication(Application application) { |
| auto conf = config_; |
| switch (application) { |
| case Application::kSpeech: |
| conf.application = AudioEncoderOpus::kVoip; |
| break; |
| case Application::kAudio: |
| conf.application = AudioEncoderOpus::kAudio; |
| break; |
| } |
| return RecreateEncoderInstance(conf); |
| } |
| |
| void AudioEncoderOpus::SetMaxPlaybackRate(int frequency_hz) { |
| auto conf = config_; |
| conf.max_playback_rate_hz = frequency_hz; |
| RTC_CHECK(RecreateEncoderInstance(conf)); |
| } |
| |
| void AudioEncoderOpus::SetProjectedPacketLossRate(double fraction) { |
| double opt_loss_rate = OptimizePacketLossRate(fraction, packet_loss_rate_); |
| if (packet_loss_rate_ != opt_loss_rate) { |
| packet_loss_rate_ = opt_loss_rate; |
| RTC_CHECK_EQ( |
| 0, WebRtcOpus_SetPacketLossRate( |
| inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5))); |
| } |
| } |
| |
| void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) { |
| config_.bitrate_bps = rtc::Optional<int>( |
| std::max(std::min(bits_per_second, kMaxBitrateBps), kMinBitrateBps)); |
| RTC_DCHECK(config_.IsOk()); |
| RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.GetBitrateBps())); |
| const auto new_complexity = config_.GetNewComplexity(); |
| if (new_complexity && complexity_ != *new_complexity) { |
| complexity_ = *new_complexity; |
| RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_)); |
| } |
| } |
| |
| bool AudioEncoderOpus::EnableAudioNetworkAdaptor( |
| const std::string& config_string, |
| const Clock* clock) { |
| audio_network_adaptor_ = audio_network_adaptor_creator_(config_string, clock); |
| return audio_network_adaptor_.get() != nullptr; |
| } |
| |
| void AudioEncoderOpus::DisableAudioNetworkAdaptor() { |
| audio_network_adaptor_.reset(nullptr); |
| } |
| |
| void AudioEncoderOpus::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) { |
| if (!audio_network_adaptor_) |
| return; |
| audio_network_adaptor_->SetUplinkBandwidth(uplink_bandwidth_bps); |
| ApplyAudioNetworkAdaptor(); |
| } |
| |
| void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction( |
| float uplink_packet_loss_fraction) { |
| if (!audio_network_adaptor_) { |
| packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction); |
| float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage(); |
| return SetProjectedPacketLossRate(average_fraction_loss); |
| } |
| audio_network_adaptor_->SetUplinkPacketLossFraction( |
| uplink_packet_loss_fraction); |
| ApplyAudioNetworkAdaptor(); |
| } |
| |
| void AudioEncoderOpus::OnReceivedTargetAudioBitrate( |
| int target_audio_bitrate_bps) { |
| if (!audio_network_adaptor_) |
| return SetTargetBitrate(target_audio_bitrate_bps); |
| audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps); |
| ApplyAudioNetworkAdaptor(); |
| } |
| |
| void AudioEncoderOpus::OnReceivedRtt(int rtt_ms) { |
| if (!audio_network_adaptor_) |
| return; |
| audio_network_adaptor_->SetRtt(rtt_ms); |
| ApplyAudioNetworkAdaptor(); |
| } |
| |
| void AudioEncoderOpus::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_); |
| |
| config_.supported_frame_lengths_ms.clear(); |
| std::copy_if(std::begin(kSupportedFrameLengths), |
| std::end(kSupportedFrameLengths), |
| std::back_inserter(config_.supported_frame_lengths_ms), |
| [&](int frame_length_ms) { |
| return frame_length_ms >= min_frame_length_ms && |
| frame_length_ms <= max_frame_length_ms; |
| }); |
| RTC_DCHECK(std::is_sorted(config_.supported_frame_lengths_ms.begin(), |
| config_.supported_frame_lengths_ms.end())); |
| } |
| |
| AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeImpl( |
| uint32_t rtp_timestamp, |
| rtc::ArrayView<const int16_t> audio, |
| rtc::Buffer* encoded) { |
| |
| 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(); |
| |
| // Will use new packet size for next encoding. |
| config_.frame_size_ms = next_frame_length_ms_; |
| |
| info.encoded_timestamp = first_timestamp_in_buffer_; |
| info.payload_type = config_.payload_type; |
| info.send_even_if_empty = true; // Allows Opus to send empty packets. |
| info.speech = (info.encoded_bytes > 0); |
| info.encoder_type = CodecType::kOpus; |
| return info; |
| } |
| |
| size_t AudioEncoderOpus::Num10msFramesPerPacket() const { |
| return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10)); |
| } |
| |
| size_t AudioEncoderOpus::SamplesPer10msFrame() const { |
| return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels; |
| } |
| |
| size_t AudioEncoderOpus::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>(config_.GetBitrateBps() / (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 AudioEncoderOpus::RecreateEncoderInstance(const Config& config) { |
| if (!config.IsOk()) |
| return false; |
| 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)); |
| RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config.GetBitrateBps())); |
| 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_ = config.GetNewComplexity().value_or(config.complexity); |
| RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_)); |
| 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))); |
| config_ = config; |
| |
| num_channels_to_encode_ = NumChannels(); |
| next_frame_length_ms_ = config_.frame_size_ms; |
| return true; |
| } |
| |
| void AudioEncoderOpus::SetFrameLength(int frame_length_ms) { |
| next_frame_length_ms_ = frame_length_ms; |
| } |
| |
| void AudioEncoderOpus::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 AudioEncoderOpus::ApplyAudioNetworkAdaptor() { |
| auto config = audio_network_adaptor_->GetEncoderRuntimeConfig(); |
| // |audio_network_adaptor_| is supposed to be configured to output all |
| // following parameters. |
| RTC_DCHECK(config.bitrate_bps); |
| RTC_DCHECK(config.frame_length_ms); |
| RTC_DCHECK(config.uplink_packet_loss_fraction); |
| RTC_DCHECK(config.enable_fec); |
| RTC_DCHECK(config.enable_dtx); |
| RTC_DCHECK(config.num_channels); |
| |
| RTC_DCHECK(*config.frame_length_ms == 20 || *config.frame_length_ms == 60); |
| |
| SetTargetBitrate(*config.bitrate_bps); |
| SetFrameLength(*config.frame_length_ms); |
| SetFec(*config.enable_fec); |
| SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction); |
| SetDtx(*config.enable_dtx); |
| SetNumChannelsToEncode(*config.num_channels); |
| } |
| |
| std::unique_ptr<AudioNetworkAdaptor> |
| AudioEncoderOpus::DefaultAudioNetworkAdaptorCreator( |
| const std::string& config_string, |
| const Clock* clock) const { |
| AudioNetworkAdaptorImpl::Config config; |
| config.clock = clock; |
| return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl( |
| config, ControllerManagerImpl::Create( |
| config_string, NumChannels(), supported_frame_lengths_ms(), |
| num_channels_to_encode_, next_frame_length_ms_, |
| GetTargetBitrate(), config_.fec_enabled, GetDtx(), clock))); |
| } |
| |
| } // namespace webrtc |