|  | /* | 
|  | *  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 MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_ | 
|  | #define MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_ | 
|  |  | 
|  | #include <memory> | 
|  | #include <string> | 
|  |  | 
|  | #include "absl/types/optional.h" | 
|  | #include "api/audio/audio_frame.h" | 
|  | #include "modules/audio_coding/neteq/audio_multi_vector.h" | 
|  | #include "modules/audio_coding/neteq/defines.h"  // Modes, Operations | 
|  | #include "modules/audio_coding/neteq/expand_uma_logger.h" | 
|  | #include "modules/audio_coding/neteq/include/neteq.h" | 
|  | #include "modules/audio_coding/neteq/packet.h" | 
|  | #include "modules/audio_coding/neteq/random_vector.h" | 
|  | #include "modules/audio_coding/neteq/statistics_calculator.h" | 
|  | #include "modules/audio_coding/neteq/tick_timer.h" | 
|  | #include "rtc_base/constructor_magic.h" | 
|  | #include "rtc_base/critical_section.h" | 
|  | #include "rtc_base/thread_annotations.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | // Forward declarations. | 
|  | class Accelerate; | 
|  | class BackgroundNoise; | 
|  | class BufferLevelFilter; | 
|  | class ComfortNoise; | 
|  | class DecisionLogic; | 
|  | class DecoderDatabase; | 
|  | class DelayManager; | 
|  | class DelayPeakDetector; | 
|  | class DtmfBuffer; | 
|  | class DtmfToneGenerator; | 
|  | class Expand; | 
|  | class Merge; | 
|  | class NackTracker; | 
|  | class Normal; | 
|  | class PacketBuffer; | 
|  | class RedPayloadSplitter; | 
|  | class PostDecodeVad; | 
|  | class PreemptiveExpand; | 
|  | class RandomVector; | 
|  | class SyncBuffer; | 
|  | class TimestampScaler; | 
|  | struct AccelerateFactory; | 
|  | struct DtmfEvent; | 
|  | struct ExpandFactory; | 
|  | struct PreemptiveExpandFactory; | 
|  |  | 
|  | class NetEqImpl : public webrtc::NetEq { | 
|  | public: | 
|  | enum class OutputType { | 
|  | kNormalSpeech, | 
|  | kPLC, | 
|  | kCNG, | 
|  | kPLCCNG, | 
|  | kVadPassive | 
|  | }; | 
|  |  | 
|  | enum ErrorCodes { | 
|  | kNoError = 0, | 
|  | kOtherError, | 
|  | kUnknownRtpPayloadType, | 
|  | kDecoderNotFound, | 
|  | kInvalidPointer, | 
|  | kAccelerateError, | 
|  | kPreemptiveExpandError, | 
|  | kComfortNoiseErrorCode, | 
|  | kDecoderErrorCode, | 
|  | kOtherDecoderError, | 
|  | kInvalidOperation, | 
|  | kDtmfParsingError, | 
|  | kDtmfInsertError, | 
|  | kSampleUnderrun, | 
|  | kDecodedTooMuch, | 
|  | kRedundancySplitError, | 
|  | kPacketBufferCorruption | 
|  | }; | 
|  |  | 
|  | struct Dependencies { | 
|  | // The constructor populates the Dependencies struct with the default | 
|  | // implementations of the objects. They can all be replaced by the user | 
|  | // before sending the struct to the NetEqImpl constructor. However, there | 
|  | // are dependencies between some of the classes inside the struct, so | 
|  | // swapping out one may make it necessary to re-create another one. | 
|  | explicit Dependencies( | 
|  | const NetEq::Config& config, | 
|  | const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory); | 
|  | ~Dependencies(); | 
|  |  | 
|  | std::unique_ptr<TickTimer> tick_timer; | 
|  | std::unique_ptr<StatisticsCalculator> stats; | 
|  | std::unique_ptr<BufferLevelFilter> buffer_level_filter; | 
|  | std::unique_ptr<DecoderDatabase> decoder_database; | 
|  | std::unique_ptr<DelayPeakDetector> delay_peak_detector; | 
|  | std::unique_ptr<DelayManager> delay_manager; | 
|  | std::unique_ptr<DtmfBuffer> dtmf_buffer; | 
|  | std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator; | 
|  | std::unique_ptr<PacketBuffer> packet_buffer; | 
|  | std::unique_ptr<RedPayloadSplitter> red_payload_splitter; | 
|  | std::unique_ptr<TimestampScaler> timestamp_scaler; | 
|  | std::unique_ptr<AccelerateFactory> accelerate_factory; | 
|  | std::unique_ptr<ExpandFactory> expand_factory; | 
|  | std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory; | 
|  | }; | 
|  |  | 
|  | // Creates a new NetEqImpl object. | 
|  | NetEqImpl(const NetEq::Config& config, | 
|  | Dependencies&& deps, | 
|  | bool create_components = true); | 
|  |  | 
|  | ~NetEqImpl() override; | 
|  |  | 
|  | // Inserts a new packet into NetEq. The |receive_timestamp| is an indication | 
|  | // of the time when the packet was received, and should be measured with | 
|  | // the same tick rate as the RTP timestamp of the current payload. | 
|  | // Returns 0 on success, -1 on failure. | 
|  | int InsertPacket(const RTPHeader& rtp_header, | 
|  | rtc::ArrayView<const uint8_t> payload, | 
|  | uint32_t receive_timestamp) override; | 
|  |  | 
|  | void InsertEmptyPacket(const RTPHeader& rtp_header) override; | 
|  |  | 
|  | int GetAudio( | 
|  | AudioFrame* audio_frame, | 
|  | bool* muted, | 
|  | absl::optional<Operations> action_override = absl::nullopt) override; | 
|  |  | 
|  | void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override; | 
|  |  | 
|  | bool RegisterPayloadType(int rtp_payload_type, | 
|  | const SdpAudioFormat& audio_format) override; | 
|  |  | 
|  | // Removes |rtp_payload_type| from the codec database. Returns 0 on success, | 
|  | // -1 on failure. | 
|  | int RemovePayloadType(uint8_t rtp_payload_type) override; | 
|  |  | 
|  | void RemoveAllPayloadTypes() override; | 
|  |  | 
|  | bool SetMinimumDelay(int delay_ms) override; | 
|  |  | 
|  | bool SetMaximumDelay(int delay_ms) override; | 
|  |  | 
|  | bool SetBaseMinimumDelayMs(int delay_ms) override; | 
|  |  | 
|  | int GetBaseMinimumDelayMs() const override; | 
|  |  | 
|  | int TargetDelayMs() const override; | 
|  |  | 
|  | int FilteredCurrentDelayMs() const override; | 
|  |  | 
|  | // Writes the current network statistics to |stats|. The statistics are reset | 
|  | // after the call. | 
|  | int NetworkStatistics(NetEqNetworkStatistics* stats) override; | 
|  |  | 
|  | NetEqLifetimeStatistics GetLifetimeStatistics() const override; | 
|  |  | 
|  | NetEqOperationsAndState GetOperationsAndState() const override; | 
|  |  | 
|  | // Enables post-decode VAD. When enabled, GetAudio() will return | 
|  | // kOutputVADPassive when the signal contains no speech. | 
|  | void EnableVad() override; | 
|  |  | 
|  | // Disables post-decode VAD. | 
|  | void DisableVad() override; | 
|  |  | 
|  | absl::optional<uint32_t> GetPlayoutTimestamp() const override; | 
|  |  | 
|  | int last_output_sample_rate_hz() const override; | 
|  |  | 
|  | absl::optional<SdpAudioFormat> GetDecoderFormat( | 
|  | int payload_type) const override; | 
|  |  | 
|  | // Flushes both the packet buffer and the sync buffer. | 
|  | void FlushBuffers() override; | 
|  |  | 
|  | void EnableNack(size_t max_nack_list_size) override; | 
|  |  | 
|  | void DisableNack() override; | 
|  |  | 
|  | std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override; | 
|  |  | 
|  | std::vector<uint32_t> LastDecodedTimestamps() const override; | 
|  |  | 
|  | int SyncBufferSizeMs() const override; | 
|  |  | 
|  | // This accessor method is only intended for testing purposes. | 
|  | const SyncBuffer* sync_buffer_for_test() const; | 
|  | Operations last_operation_for_test() const; | 
|  |  | 
|  | protected: | 
|  | static const int kOutputSizeMs = 10; | 
|  | static const size_t kMaxFrameSize = 5760;  // 120 ms @ 48 kHz. | 
|  | // TODO(hlundin): Provide a better value for kSyncBufferSize. | 
|  | // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for | 
|  | // calculating correlations of current frame against history. | 
|  | static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48; | 
|  |  | 
|  | // Inserts a new packet into NetEq. This is used by the InsertPacket method | 
|  | // above. Returns 0 on success, otherwise an error code. | 
|  | // TODO(hlundin): Merge this with InsertPacket above? | 
|  | int InsertPacketInternal(const RTPHeader& rtp_header, | 
|  | rtc::ArrayView<const uint8_t> payload, | 
|  | uint32_t receive_timestamp) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Delivers 10 ms of audio data. The data is written to |audio_frame|. | 
|  | // Returns 0 on success, otherwise an error code. | 
|  | int GetAudioInternal(AudioFrame* audio_frame, | 
|  | bool* muted, | 
|  | absl::optional<Operations> action_override) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Provides a decision to the GetAudioInternal method. The decision what to | 
|  | // do is written to |operation|. Packets to decode are written to | 
|  | // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When | 
|  | // DTMF should be played, |play_dtmf| is set to true by the method. | 
|  | // Returns 0 on success, otherwise an error code. | 
|  | int GetDecision(Operations* operation, | 
|  | PacketList* packet_list, | 
|  | DtmfEvent* dtmf_event, | 
|  | bool* play_dtmf, | 
|  | absl::optional<Operations> action_override) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Decodes the speech packets in |packet_list|, and writes the results to | 
|  | // |decoded_buffer|, which is allocated to hold |decoded_buffer_length| | 
|  | // elements. The length of the decoded data is written to |decoded_length|. | 
|  | // The speech type -- speech or (codec-internal) comfort noise -- is written | 
|  | // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389 | 
|  | // comfort noise, those are not decoded. | 
|  | int Decode(PacketList* packet_list, | 
|  | Operations* operation, | 
|  | int* decoded_length, | 
|  | AudioDecoder::SpeechType* speech_type) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Sub-method to Decode(). Performs codec internal CNG. | 
|  | int DecodeCng(AudioDecoder* decoder, | 
|  | int* decoded_length, | 
|  | AudioDecoder::SpeechType* speech_type) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Sub-method to Decode(). Performs the actual decoding. | 
|  | int DecodeLoop(PacketList* packet_list, | 
|  | const Operations& operation, | 
|  | AudioDecoder* decoder, | 
|  | int* decoded_length, | 
|  | AudioDecoder::SpeechType* speech_type) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Sub-method which calls the Normal class to perform the normal operation. | 
|  | void DoNormal(const int16_t* decoded_buffer, | 
|  | size_t decoded_length, | 
|  | AudioDecoder::SpeechType speech_type, | 
|  | bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Sub-method which calls the Merge class to perform the merge operation. | 
|  | void DoMerge(int16_t* decoded_buffer, | 
|  | size_t decoded_length, | 
|  | AudioDecoder::SpeechType speech_type, | 
|  | bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | bool DoCodecPlc() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Sub-method which calls the Expand class to perform the expand operation. | 
|  | int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Sub-method which calls the Accelerate class to perform the accelerate | 
|  | // operation. | 
|  | int DoAccelerate(int16_t* decoded_buffer, | 
|  | size_t decoded_length, | 
|  | AudioDecoder::SpeechType speech_type, | 
|  | bool play_dtmf, | 
|  | bool fast_accelerate) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Sub-method which calls the PreemptiveExpand class to perform the | 
|  | // preemtive expand operation. | 
|  | int DoPreemptiveExpand(int16_t* decoded_buffer, | 
|  | size_t decoded_length, | 
|  | AudioDecoder::SpeechType speech_type, | 
|  | bool play_dtmf) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort | 
|  | // noise. |packet_list| can either contain one SID frame to update the | 
|  | // noise parameters, or no payload at all, in which case the previously | 
|  | // received parameters are used. | 
|  | int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Calls the audio decoder to generate codec-internal comfort noise when | 
|  | // no packet was received. | 
|  | void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Calls the DtmfToneGenerator class to generate DTMF tones. | 
|  | int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Overdub DTMF on top of |output|. | 
|  | int DtmfOverdub(const DtmfEvent& dtmf_event, | 
|  | size_t num_channels, | 
|  | int16_t* output) const | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Extracts packets from |packet_buffer_| to produce at least | 
|  | // |required_samples| samples. The packets are inserted into |packet_list|. | 
|  | // Returns the number of samples that the packets in the list will produce, or | 
|  | // -1 in case of an error. | 
|  | int ExtractPackets(size_t required_samples, PacketList* packet_list) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Resets various variables and objects to new values based on the sample rate | 
|  | // |fs_hz| and |channels| number audio channels. | 
|  | void SetSampleRateAndChannels(int fs_hz, size_t channels) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Returns the output type for the audio produced by the latest call to | 
|  | // GetAudio(). | 
|  | OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Updates Expand and Merge. | 
|  | virtual void UpdatePlcComponents(int fs_hz, size_t channels) | 
|  | RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | // Creates DecisionLogic object with the mode given by |playout_mode_|. | 
|  | virtual void CreateDecisionLogic() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); | 
|  |  | 
|  | rtc::CriticalSection crit_sect_; | 
|  | const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<BufferLevelFilter> buffer_level_filter_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<DecoderDatabase> decoder_database_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<DelayManager> delay_manager_ RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<DelayPeakDetector> delay_peak_detector_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<TimestampScaler> timestamp_scaler_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<ExpandFactory> expand_factory_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<AccelerateFactory> accelerate_factory_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | const std::unique_ptr<StatisticsCalculator> stats_ RTC_GUARDED_BY(crit_sect_); | 
|  |  | 
|  | std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<DecisionLogic> decision_logic_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<AudioMultiVector> algorithm_buffer_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<PreemptiveExpand> preemptive_expand_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_); | 
|  | int fs_hz_ RTC_GUARDED_BY(crit_sect_); | 
|  | int fs_mult_ RTC_GUARDED_BY(crit_sect_); | 
|  | int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_); | 
|  | size_t output_size_samples_ RTC_GUARDED_BY(crit_sect_); | 
|  | size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_); | 
|  | Modes last_mode_ RTC_GUARDED_BY(crit_sect_); | 
|  | Operations last_operation_ RTC_GUARDED_BY(crit_sect_); | 
|  | size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_); | 
|  | uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_); | 
|  | bool new_codec_ RTC_GUARDED_BY(crit_sect_); | 
|  | uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_); | 
|  | bool reset_decoder_ RTC_GUARDED_BY(crit_sect_); | 
|  | absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_); | 
|  | absl::optional<uint8_t> current_cng_rtp_payload_type_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | bool first_packet_ RTC_GUARDED_BY(crit_sect_); | 
|  | bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_); | 
|  | std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_); | 
|  | bool nack_enabled_ RTC_GUARDED_BY(crit_sect_); | 
|  | const bool enable_muted_state_ RTC_GUARDED_BY(crit_sect_); | 
|  | AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(crit_sect_) = | 
|  | AudioFrame::kVadPassive; | 
|  | std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_ | 
|  | RTC_GUARDED_BY(crit_sect_); | 
|  | std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_); | 
|  | ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_); | 
|  | ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_); | 
|  | bool no_time_stretching_ RTC_GUARDED_BY(crit_sect_);  // Only used for test. | 
|  | rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(crit_sect_); | 
|  | const bool enable_rtx_handling_ RTC_GUARDED_BY(crit_sect_); | 
|  |  | 
|  | private: | 
|  | RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl); | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  | #endif  // MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_ |