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Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.h')
| -rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.h | 360 |
1 files changed, 0 insertions, 360 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.h b/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.h deleted file mode 100644 index 83dd58b5acb..00000000000 --- a/chromium/third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.h +++ /dev/null @@ -1,360 +0,0 @@ -/* - * 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NETEQ_IMPL_H_ -#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NETEQ_IMPL_H_ - -#include <vector> - -#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" -#include "webrtc/modules/audio_coding/neteq4/defines.h" -#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h" -#include "webrtc/modules/audio_coding/neteq4/packet.h" // Declare PacketList. -#include "webrtc/modules/audio_coding/neteq4/random_vector.h" -#include "webrtc/modules/audio_coding/neteq4/rtcp.h" -#include "webrtc/modules/audio_coding/neteq4/statistics_calculator.h" -#include "webrtc/system_wrappers/interface/constructor_magic.h" -#include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -// Forward declarations. -class Accelerate; -class BackgroundNoise; -class BufferLevelFilter; -class ComfortNoise; -class CriticalSectionWrapper; -class DecisionLogic; -class DecoderDatabase; -class DelayManager; -class DelayPeakDetector; -class DtmfBuffer; -class DtmfToneGenerator; -class Expand; -class Merge; -class Normal; -class PacketBuffer; -class PayloadSplitter; -class PostDecodeVad; -class PreemptiveExpand; -class RandomVector; -class SyncBuffer; -class TimestampScaler; -struct DtmfEvent; - -class NetEqImpl : public webrtc::NetEq { - public: - // Creates a new NetEqImpl object. The object will assume ownership of all - // injected dependencies, and will delete them when done. - NetEqImpl(int fs, - BufferLevelFilter* buffer_level_filter, - DecoderDatabase* decoder_database, - DelayManager* delay_manager, - DelayPeakDetector* delay_peak_detector, - DtmfBuffer* dtmf_buffer, - DtmfToneGenerator* dtmf_tone_generator, - PacketBuffer* packet_buffer, - PayloadSplitter* payload_splitter, - TimestampScaler* timestamp_scaler); - - virtual ~NetEqImpl(); - - // 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. - virtual int InsertPacket(const WebRtcRTPHeader& rtp_header, - const uint8_t* payload, - int length_bytes, - uint32_t receive_timestamp); - - // Inserts a sync-packet into packet queue. Sync-packets are decoded to - // silence and are intended to keep AV-sync intact in an event of long packet - // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq - // might insert sync-packet when they observe that buffer level of NetEq is - // decreasing below a certain threshold, defined by the application. - // Sync-packets should have the same payload type as the last audio payload - // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change - // can be implied by inserting a sync-packet. - // Returns kOk on success, kFail on failure. - virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header, - uint32_t receive_timestamp); - - // Instructs NetEq to deliver 10 ms of audio data. The data is written to - // |output_audio|, which can hold (at least) |max_length| elements. - // The number of channels that were written to the output is provided in - // the output variable |num_channels|, and each channel contains - // |samples_per_channel| elements. If more than one channel is written, - // the samples are interleaved. - // The speech type is written to |type|, if |type| is not NULL. - // Returns kOK on success, or kFail in case of an error. - virtual int GetAudio(size_t max_length, int16_t* output_audio, - int* samples_per_channel, int* num_channels, - NetEqOutputType* type); - - // Associates |rtp_payload_type| with |codec| and stores the information in - // the codec database. Returns kOK on success, kFail on failure. - virtual int RegisterPayloadType(enum NetEqDecoder codec, - uint8_t rtp_payload_type); - - // Provides an externally created decoder object |decoder| to insert in the - // decoder database. The decoder implements a decoder of type |codec| and - // associates it with |rtp_payload_type|. The decoder operates at the - // frequency |sample_rate_hz|. Returns kOK on success, kFail on failure. - virtual int RegisterExternalDecoder(AudioDecoder* decoder, - enum NetEqDecoder codec, - int sample_rate_hz, - uint8_t rtp_payload_type); - - // Removes |rtp_payload_type| from the codec database. Returns 0 on success, - // -1 on failure. - virtual int RemovePayloadType(uint8_t rtp_payload_type); - - virtual bool SetMinimumDelay(int delay_ms); - - virtual bool SetMaximumDelay(int delay_ms); - - virtual int LeastRequiredDelayMs() const; - - virtual int SetTargetDelay() { return kNotImplemented; } - - virtual int TargetDelay() { return kNotImplemented; } - - virtual int CurrentDelay() { return kNotImplemented; } - - // Sets the playout mode to |mode|. - virtual void SetPlayoutMode(NetEqPlayoutMode mode); - - // Returns the current playout mode. - virtual NetEqPlayoutMode PlayoutMode() const; - - // Writes the current network statistics to |stats|. The statistics are reset - // after the call. - virtual int NetworkStatistics(NetEqNetworkStatistics* stats); - - // Writes the last packet waiting times (in ms) to |waiting_times|. The number - // of values written is no more than 100, but may be smaller if the interface - // is polled again before 100 packets has arrived. - virtual void WaitingTimes(std::vector<int>* waiting_times); - - // Writes the current RTCP statistics to |stats|. The statistics are reset - // and a new report period is started with the call. - virtual void GetRtcpStatistics(RtcpStatistics* stats); - - // Same as RtcpStatistics(), but does not reset anything. - virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats); - - // Enables post-decode VAD. When enabled, GetAudio() will return - // kOutputVADPassive when the signal contains no speech. - virtual void EnableVad(); - - // Disables post-decode VAD. - virtual void DisableVad(); - - // Returns the RTP timestamp for the last sample delivered by GetAudio(). - virtual uint32_t PlayoutTimestamp(); - - virtual int SetTargetNumberOfChannels() { return kNotImplemented; } - - virtual int SetTargetSampleRate() { return kNotImplemented; } - - // Returns the error code for the last occurred error. If no error has - // occurred, 0 is returned. - virtual int LastError(); - - // Returns the error code last returned by a decoder (audio or comfort noise). - // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check - // this method to get the decoder's error code. - virtual int LastDecoderError(); - - // Flushes both the packet buffer and the sync buffer. - virtual void FlushBuffers(); - - virtual void PacketBufferStatistics(int* current_num_packets, - int* max_num_packets, - int* current_memory_size_bytes, - int* max_memory_size_bytes) const; - - // Get sequence number and timestamp of the latest RTP. - // This method is to facilitate NACK. - virtual int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const; - - // Sets background noise mode. - virtual void SetBackgroundNoiseMode(NetEqBackgroundNoiseMode mode); - - // Gets background noise mode. - virtual NetEqBackgroundNoiseMode BackgroundNoiseMode() const; - - private: - static const int kOutputSizeMs = 10; - static const int kMaxFrameSize = 2880; // 60 ms @ 48 kHz. - // TODO(hlundin): Provide a better value for kSyncBufferSize. - static const int kSyncBufferSize = 2 * kMaxFrameSize; - - // 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 WebRtcRTPHeader& rtp_header, - const uint8_t* payload, - int length_bytes, - uint32_t receive_timestamp, - bool is_sync_packet); - - - // Delivers 10 ms of audio data. The data is written to |output|, which can - // hold (at least) |max_length| elements. The number of channels that were - // written to the output is provided in the output variable |num_channels|, - // and each channel contains |samples_per_channel| elements. If more than one - // channel is written, the samples are interleaved. - // Returns 0 on success, otherwise an error code. - int GetAudioInternal(size_t max_length, int16_t* output, - int* samples_per_channel, int* num_channels); - - - // 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); - - // 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); - - // Sub-method to Decode(). Performs the actual decoding. - int DecodeLoop(PacketList* packet_list, Operations* operation, - AudioDecoder* decoder, int* decoded_length, - AudioDecoder::SpeechType* speech_type); - - // 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); - - // 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); - - // Sub-method which calls the Expand class to perform the expand operation. - int DoExpand(bool play_dtmf); - - // 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); - - // 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); - - // 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); - - // Calls the audio decoder to generate codec-internal comfort noise when - // no packet was received. - void DoCodecInternalCng(); - - // Calls the DtmfToneGenerator class to generate DTMF tones. - int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf); - - // Produces packet-loss concealment using alternative methods. If the codec - // has an internal PLC, it is called to generate samples. Otherwise, the - // method performs zero-stuffing. - void DoAlternativePlc(bool increase_timestamp); - - // Overdub DTMF on top of |output|. - int DtmfOverdub(const DtmfEvent& dtmf_event, size_t num_channels, - int16_t* output) const; - - // 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(int required_samples, PacketList* packet_list); - - // 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); - - // Returns the output type for the audio produced by the latest call to - // GetAudio(). - NetEqOutputType LastOutputType(); - - scoped_ptr<BackgroundNoise> background_noise_; - scoped_ptr<BufferLevelFilter> buffer_level_filter_; - scoped_ptr<DecoderDatabase> decoder_database_; - scoped_ptr<DelayManager> delay_manager_; - scoped_ptr<DelayPeakDetector> delay_peak_detector_; - scoped_ptr<DtmfBuffer> dtmf_buffer_; - scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_; - scoped_ptr<PacketBuffer> packet_buffer_; - scoped_ptr<PayloadSplitter> payload_splitter_; - scoped_ptr<TimestampScaler> timestamp_scaler_; - scoped_ptr<DecisionLogic> decision_logic_; - scoped_ptr<PostDecodeVad> vad_; - scoped_ptr<AudioMultiVector> algorithm_buffer_; - scoped_ptr<SyncBuffer> sync_buffer_; - scoped_ptr<Expand> expand_; - scoped_ptr<Normal> normal_; - scoped_ptr<Merge> merge_; - scoped_ptr<Accelerate> accelerate_; - scoped_ptr<PreemptiveExpand> preemptive_expand_; - RandomVector random_vector_; - scoped_ptr<ComfortNoise> comfort_noise_; - Rtcp rtcp_; - StatisticsCalculator stats_; - int fs_hz_; - int fs_mult_; - int output_size_samples_; - int decoder_frame_length_; - Modes last_mode_; - scoped_array<int16_t> mute_factor_array_; - size_t decoded_buffer_length_; - scoped_array<int16_t> decoded_buffer_; - uint32_t playout_timestamp_; - bool new_codec_; - uint32_t timestamp_; - bool reset_decoder_; - uint8_t current_rtp_payload_type_; - uint8_t current_cng_rtp_payload_type_; - uint32_t ssrc_; - bool first_packet_; - int error_code_; // Store last error code. - int decoder_error_code_; - scoped_ptr<CriticalSectionWrapper> crit_sect_; - - // These values are used by NACK module to estimate time-to-play of - // a missing packet. Occasionally, NetEq might decide to decode more - // than one packet. Therefore, these values store sequence number and - // timestamp of the first packet pulled from the packet buffer. In - // such cases, these values do not exactly represent the sequence number - // or timestamp associated with a 10ms audio pulled from NetEq. NACK - // module is designed to compensate for this. - int decoded_packet_sequence_number_; - uint32_t decoded_packet_timestamp_; - - DISALLOW_COPY_AND_ASSIGN(NetEqImpl); -}; - -} // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NETEQ_IMPL_H_ |