| /* |
| * 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_MAIN_SOURCE_ACM_NETEQ_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ |
| |
| #include "webrtc/common_audio/vad/include/webrtc_vad.h" |
| #include "webrtc/engine_configurations.h" |
| #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" |
| #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" |
| #include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h" |
| #include "webrtc/modules/interface/module_common_types.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| class CriticalSectionWrapper; |
| class RWLockWrapper; |
| struct CodecInst; |
| |
| #define MAX_NUM_SLAVE_NETEQ 1 |
| |
| class ACMNetEQ { |
| public: |
| enum JitterBuffer { |
| kMasterJb = 0, |
| kSlaveJb = 1 |
| }; |
| |
| // Constructor of the class |
| ACMNetEQ(); |
| |
| // Destructor of the class. |
| ~ACMNetEQ(); |
| |
| // |
| // Init() |
| // Allocates memory for NetEQ and VAD and initializes them. |
| // |
| // Return value : 0 if ok. |
| // -1 if NetEQ or VAD returned an error or |
| // if out of memory. |
| // |
| WebRtc_Word32 Init(); |
| |
| // |
| // RecIn() |
| // Gives the payload to NetEQ. |
| // |
| // Input: |
| // - incoming_payload : Incoming audio payload. |
| // - length_payload : Length of incoming audio payload. |
| // - rtp_info : RTP header for the incoming payload containing |
| // information about payload type, sequence number, |
| // timestamp, SSRC and marker bit. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 RecIn(const WebRtc_UWord8* incoming_payload, |
| const WebRtc_Word32 length_payload, |
| const WebRtcRTPHeader& rtp_info); |
| |
| // |
| // RecOut() |
| // Asks NetEQ for 10 ms of decoded audio. |
| // |
| // Input: |
| // -audio_frame : an audio frame were output data and |
| // associated parameters are written to. |
| // |
| // Return value : 0 if ok. |
| // -1 if NetEQ returned an error. |
| // |
| WebRtc_Word32 RecOut(AudioFrame& audio_frame); |
| |
| // |
| // AddCodec() |
| // Adds a new codec to the NetEQ codec database. |
| // |
| // Input: |
| // - codec_def : The codec to be added. |
| // - to_master : true if the codec has to be added to Master |
| // NetEq, otherwise will be added to the Slave |
| // NetEQ. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 AddCodec(WebRtcNetEQ_CodecDef *codec_def, |
| bool to_master = true); |
| |
| // |
| // AllocatePacketBuffer() |
| // Allocates the NetEQ packet buffer. |
| // |
| // Input: |
| // - used_codecs : An array of the codecs to be used by NetEQ. |
| // - num_codecs : Number of codecs in used_codecs. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 AllocatePacketBuffer(const WebRtcNetEQDecoder* used_codecs, |
| WebRtc_Word16 num_codecs); |
| |
| // |
| // SetExtraDelay() |
| // Sets a |delay_in_ms| milliseconds extra delay in NetEQ. |
| // |
| // Input: |
| // - delay_in_ms : Extra delay in milliseconds. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 SetExtraDelay(const WebRtc_Word32 delay_in_ms); |
| |
| // |
| // SetAVTPlayout() |
| // Enable/disable playout of AVT payloads. |
| // |
| // Input: |
| // - enable : Enable if true, disable if false. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 SetAVTPlayout(const bool enable); |
| |
| // |
| // AVTPlayout() |
| // Get the current AVT playout state. |
| // |
| // Return value : True if AVT playout is enabled. |
| // False if AVT playout is disabled. |
| // |
| bool avt_playout() const; |
| |
| // |
| // CurrentSampFreqHz() |
| // Get the current sampling frequency in Hz. |
| // |
| // Return value : Sampling frequency in Hz. |
| // |
| WebRtc_Word32 CurrentSampFreqHz() const; |
| |
| // |
| // SetPlayoutMode() |
| // Sets the playout mode to voice or fax. |
| // |
| // Input: |
| // - mode : The playout mode to be used, voice, |
| // fax, or streaming. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 SetPlayoutMode(const AudioPlayoutMode mode); |
| |
| // |
| // PlayoutMode() |
| // Get the current playout mode. |
| // |
| // Return value : The current playout mode. |
| // |
| AudioPlayoutMode playout_mode() const; |
| |
| // |
| // NetworkStatistics() |
| // Get the current network statistics from NetEQ. |
| // |
| // Output: |
| // - statistics : The current network statistics. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| WebRtc_Word32 NetworkStatistics(ACMNetworkStatistics* statistics) const; |
| |
| // |
| // VADMode() |
| // Get the current VAD Mode. |
| // |
| // Return value : The current VAD mode. |
| // |
| ACMVADMode vad_mode() const; |
| |
| // |
| // SetVADMode() |
| // Set the VAD mode. |
| // |
| // Input: |
| // - mode : The new VAD mode. |
| // |
| // Return value : 0 if ok. |
| // -1 if an error occurred. |
| // |
| WebRtc_Word16 SetVADMode(const ACMVADMode mode); |
| |
| // |
| // DecodeLock() |
| // Get the decode lock used to protect decoder instances while decoding. |
| // |
| // Return value : Pointer to the decode lock. |
| // |
| RWLockWrapper* DecodeLock() const { |
| return decode_lock_; |
| } |
| |
| // |
| // FlushBuffers() |
| // Flushes the NetEQ packet and speech buffers. |
| // |
| // Return value : 0 if ok. |
| // -1 if NetEQ returned an error. |
| // |
| WebRtc_Word32 FlushBuffers(); |
| |
| // |
| // RemoveCodec() |
| // Removes a codec from the NetEQ codec database. |
| // |
| // Input: |
| // - codec_idx : Codec to be removed. |
| // |
| // Return value : 0 if ok. |
| // -1 if an error occurred. |
| // |
| WebRtc_Word16 RemoveCodec(WebRtcNetEQDecoder codec_idx, |
| bool is_stereo = false); |
| |
| // |
| // SetBackgroundNoiseMode() |
| // Set the mode of the background noise. |
| // |
| // Input: |
| // - mode : an enumerator specifying the mode of the |
| // background noise. |
| // |
| // Return value : 0 if succeeded, |
| // -1 if failed to set the mode. |
| // |
| WebRtc_Word16 SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode); |
| |
| // |
| // BackgroundNoiseMode() |
| // return the mode of the background noise. |
| // |
| // Return value : The mode of background noise. |
| // |
| WebRtc_Word16 BackgroundNoiseMode(ACMBackgroundNoiseMode& mode); |
| |
| void set_id(WebRtc_Word32 id); |
| |
| WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32& timestamp); |
| |
| void set_received_stereo(bool received_stereo); |
| |
| WebRtc_UWord8 num_slaves(); |
| |
| // Delete all slaves. |
| void RemoveSlaves(); |
| |
| WebRtc_Word16 AddSlave(const WebRtcNetEQDecoder* used_codecs, |
| WebRtc_Word16 num_codecs); |
| |
| private: |
| // |
| // RTPPack() |
| // Creates a Word16 RTP packet out of the payload data in Word16 and |
| // a WebRtcRTPHeader. |
| // |
| // Input: |
| // - payload : Payload to be packetized. |
| // - payload_length_bytes : Length of the payload in bytes. |
| // - rtp_info : RTP header structure. |
| // |
| // Output: |
| // - rtp_packet : The RTP packet. |
| // |
| static void RTPPack(WebRtc_Word16* rtp_packet, const WebRtc_Word8* payload, |
| const WebRtc_Word32 payload_length_bytes, |
| const WebRtcRTPHeader& rtp_info); |
| |
| void LogError(const char* neteq_func_name, const WebRtc_Word16 idx) const; |
| |
| WebRtc_Word16 InitByIdxSafe(const WebRtc_Word16 idx); |
| |
| // |
| // EnableVAD() |
| // Enable VAD. |
| // |
| // Return value : 0 if ok. |
| // -1 if an error occurred. |
| // |
| WebRtc_Word16 EnableVAD(); |
| |
| WebRtc_Word16 EnableVADByIdxSafe(const WebRtc_Word16 idx); |
| |
| WebRtc_Word16 AllocatePacketBufferByIdxSafe( |
| const WebRtcNetEQDecoder* used_codecs, |
| WebRtc_Word16 num_codecs, |
| const WebRtc_Word16 idx); |
| |
| // Delete the NetEQ corresponding to |index|. |
| void RemoveNetEQSafe(int index); |
| |
| void RemoveSlavesSafe(); |
| |
| void* inst_[MAX_NUM_SLAVE_NETEQ + 1]; |
| void* inst_mem_[MAX_NUM_SLAVE_NETEQ + 1]; |
| |
| WebRtc_Word16* neteq_packet_buffer_[MAX_NUM_SLAVE_NETEQ + 1]; |
| |
| WebRtc_Word32 id_; |
| float current_samp_freq_khz_; |
| bool avt_playout_; |
| AudioPlayoutMode playout_mode_; |
| CriticalSectionWrapper* neteq_crit_sect_; |
| |
| WebRtcVadInst* ptr_vadinst_[MAX_NUM_SLAVE_NETEQ + 1]; |
| |
| bool vad_status_; |
| ACMVADMode vad_mode_; |
| RWLockWrapper* decode_lock_; |
| bool is_initialized_[MAX_NUM_SLAVE_NETEQ + 1]; |
| WebRtc_UWord8 num_slaves_; |
| bool received_stereo_; |
| void* master_slave_info_; |
| AudioFrame::VADActivity previous_audio_activity_; |
| WebRtc_Word32 extra_delay_; |
| |
| CriticalSectionWrapper* callback_crit_sect_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ |
