| /* |
| * 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ |
| |
| #include <functional> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/base/optional.h" |
| #include "webrtc/common_audio/smoothing_filter.h" |
| #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" |
| #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" |
| #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
| |
| namespace webrtc { |
| |
| class RtcEventLog; |
| |
| struct CodecInst; |
| |
| class AudioEncoderOpus final : public AudioEncoder { |
| public: |
| enum ApplicationMode { |
| kVoip = 0, |
| kAudio = 1, |
| }; |
| |
| struct Config { |
| Config(); |
| Config(const Config&); |
| ~Config(); |
| Config& operator=(const Config&); |
| |
| bool IsOk() const; |
| int GetBitrateBps() const; |
| // Returns empty if the current bitrate falls within the hysteresis window, |
| // defined by complexity_threshold_bps +/- complexity_threshold_window_bps. |
| // Otherwise, returns the current complexity depending on whether the |
| // current bitrate is above or below complexity_threshold_bps. |
| rtc::Optional<int> GetNewComplexity() const; |
| |
| int frame_size_ms = 20; |
| size_t num_channels = 1; |
| int payload_type = 120; |
| ApplicationMode application = kVoip; |
| rtc::Optional<int> bitrate_bps; // Unset means to use default value. |
| bool fec_enabled = false; |
| int max_playback_rate_hz = 48000; |
| int complexity = kDefaultComplexity; |
| // This value may change in the struct's constructor. |
| int low_rate_complexity = kDefaultComplexity; |
| // low_rate_complexity is used when the bitrate is below this threshold. |
| int complexity_threshold_bps = 12500; |
| int complexity_threshold_window_bps = 1500; |
| bool dtx_enabled = false; |
| std::vector<int> supported_frame_lengths_ms; |
| const Clock* clock = Clock::GetRealTimeClock(); |
| int uplink_bandwidth_update_interval_ms = 200; |
| |
| private: |
| #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM) |
| // If we are on Android, iOS and/or ARM, use a lower complexity setting as |
| // default, to save encoder complexity. |
| static const int kDefaultComplexity = 5; |
| #else |
| static const int kDefaultComplexity = 9; |
| #endif |
| }; |
| |
| using AudioNetworkAdaptorCreator = |
| std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&, |
| RtcEventLog*, |
| const Clock*)>; |
| AudioEncoderOpus( |
| const Config& config, |
| AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr, |
| std::unique_ptr<SmoothingFilter> bitrate_smoother = nullptr); |
| |
| explicit AudioEncoderOpus(const CodecInst& codec_inst); |
| |
| ~AudioEncoderOpus() 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 Reset() override; |
| bool SetFec(bool enable) override; |
| |
| // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice |
| // being inactive. During that, it still sends 2 packets (one for content, one |
| // for signaling) about every 400 ms. |
| bool SetDtx(bool enable) override; |
| bool GetDtx() const override; |
| |
| bool SetApplication(Application application) override; |
| void SetMaxPlaybackRate(int frequency_hz) override; |
| bool EnableAudioNetworkAdaptor(const std::string& config_string, |
| RtcEventLog* event_log, |
| const Clock* clock) override; |
| void DisableAudioNetworkAdaptor() override; |
| void OnReceivedUplinkPacketLossFraction( |
| float uplink_packet_loss_fraction) override; |
| void OnReceivedUplinkRecoverablePacketLossFraction( |
| float uplink_recoverable_packet_loss_fraction) override; |
| void OnReceivedUplinkBandwidth( |
| int target_audio_bitrate_bps, |
| rtc::Optional<int64_t> probing_interval_ms) override; |
| void OnReceivedRtt(int rtt_ms) override; |
| void OnReceivedOverhead(size_t overhead_bytes_per_packet) override; |
| void SetReceiverFrameLengthRange(int min_frame_length_ms, |
| int max_frame_length_ms) override; |
| rtc::ArrayView<const int> supported_frame_lengths_ms() const { |
| return config_.supported_frame_lengths_ms; |
| } |
| |
| // Getters for testing. |
| float packet_loss_rate() const { return packet_loss_rate_; } |
| ApplicationMode application() const { return config_.application; } |
| bool fec_enabled() const { return config_.fec_enabled; } |
| size_t num_channels_to_encode() const { return num_channels_to_encode_; } |
| int next_frame_length_ms() const { return next_frame_length_ms_; } |
| |
| protected: |
| EncodedInfo EncodeImpl(uint32_t rtp_timestamp, |
| rtc::ArrayView<const int16_t> audio, |
| rtc::Buffer* encoded) override; |
| |
| private: |
| class PacketLossFractionSmoother; |
| |
| size_t Num10msFramesPerPacket() const; |
| size_t SamplesPer10msFrame() const; |
| size_t SufficientOutputBufferSize() const; |
| bool RecreateEncoderInstance(const Config& config); |
| void SetFrameLength(int frame_length_ms); |
| void SetNumChannelsToEncode(size_t num_channels_to_encode); |
| void SetProjectedPacketLossRate(float fraction); |
| |
| // TODO(minyue): remove "override" when we can deprecate |
| // |AudioEncoder::SetTargetBitrate|. |
| void SetTargetBitrate(int target_bps) override; |
| |
| void ApplyAudioNetworkAdaptor(); |
| std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator( |
| const std::string& config_string, |
| RtcEventLog* event_log, |
| const Clock* clock) const; |
| |
| void MaybeUpdateUplinkBandwidth(); |
| |
| Config config_; |
| const bool send_side_bwe_with_overhead_; |
| float packet_loss_rate_; |
| std::vector<int16_t> input_buffer_; |
| OpusEncInst* inst_; |
| uint32_t first_timestamp_in_buffer_; |
| size_t num_channels_to_encode_; |
| int next_frame_length_ms_; |
| int complexity_; |
| std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_; |
| AudioNetworkAdaptorCreator audio_network_adaptor_creator_; |
| std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_; |
| rtc::Optional<size_t> overhead_bytes_per_packet_; |
| const std::unique_ptr<SmoothingFilter> bitrate_smoother_; |
| rtc::Optional<int64_t> bitrate_smoother_last_update_time_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ |