| /* |
| * 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/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; |
| |
| namespace acm1 { |
| |
| #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. |
| // |
| int32_t 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. |
| // - receive_timestamp : received timestamp. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| int32_t RecIn(const uint8_t* incoming_payload, |
| const int32_t length_payload, |
| const WebRtcRTPHeader& rtp_info, |
| uint32_t receive_timestamp); |
| |
| // |
| // RecIn() |
| // Insert a sync payload to NetEq. Should only be called if |av_sync_| is |
| // enabled; |
| // |
| // Input: |
| // - rtp_info : RTP header for the incoming payload containing |
| // information about payload type, sequence number, |
| // timestamp, SSRC and marker bit. |
| // - receive_timestamp : received timestamp. |
| // |
| // Return value : 0 if ok. |
| // <0 if NetEQ returned an error. |
| // |
| int RecIn(const WebRtcRTPHeader& rtp_info, uint32_t receive_timestamp); |
| |
| // |
| // 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. |
| // |
| int32_t 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. |
| // |
| int32_t 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. |
| // |
| int32_t AllocatePacketBuffer(const WebRtcNetEQDecoder* used_codecs, |
| int16_t num_codecs); |
| |
| // |
| // 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. |
| // |
| int32_t 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. |
| // |
| int32_t 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. |
| // |
| int32_t 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. |
| // |
| int32_t 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. |
| // |
| int16_t 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. |
| // |
| int32_t 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. |
| // |
| int16_t 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. |
| // |
| int16_t SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode); |
| |
| // |
| // BackgroundNoiseMode() |
| // return the mode of the background noise. |
| // |
| // Return value : The mode of background noise. |
| // |
| int16_t BackgroundNoiseMode(ACMBackgroundNoiseMode& mode); |
| |
| void set_id(int32_t id); |
| |
| int32_t PlayoutTimestamp(uint32_t& timestamp); |
| |
| void set_received_stereo(bool received_stereo); |
| |
| uint8_t num_slaves(); |
| |
| // Delete all slaves. |
| void RemoveSlaves(); |
| |
| int16_t AddSlave(const WebRtcNetEQDecoder* used_codecs, |
| int16_t num_codecs); |
| |
| void BufferSpec(int& num_packets, int& size_bytes, int& overhead_bytes) { |
| num_packets = min_of_max_num_packets_; |
| size_bytes = min_of_buffer_size_bytes_; |
| overhead_bytes = per_packet_overhead_bytes_; |
| } |
| |
| // |
| // Set AV-sync mode. |
| // |
| void EnableAVSync(bool enable); |
| |
| // |
| // Get sequence number and timestamp of the last decoded RTP. |
| // |
| bool DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const; |
| |
| // |
| // Set a minimum delay in NetEq. Unless channel condition dictates a longer |
| // delay, the given delay is maintained by NetEq. |
| // |
| int SetMinimumDelay(int minimum_delay_ms); |
| |
| // |
| // Set a maximum delay in NetEq. |
| // |
| int SetMaximumDelay(int maximum_delay_ms); |
| |
| // |
| // The shortest latency, in milliseconds, required by jitter buffer. This |
| // is computed based on inter-arrival times and playout mode of NetEq. The |
| // actual delay is the maximum of least-required-delay and the minimum-delay |
| // specified by SetMinumumPlayoutDelay() API. |
| // |
| int LeastRequiredDelayMs() const ; |
| |
| 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(int16_t* rtp_packet, const int8_t* payload, |
| const int32_t payload_length_bytes, |
| const WebRtcRTPHeader& rtp_info); |
| |
| void LogError(const char* neteq_func_name, const int16_t idx) const; |
| |
| int16_t InitByIdxSafe(const int16_t idx); |
| |
| // |
| // EnableVAD() |
| // Enable VAD. |
| // |
| // Return value : 0 if ok. |
| // -1 if an error occurred. |
| // |
| int16_t EnableVAD(); |
| |
| int16_t EnableVADByIdxSafe(const int16_t idx); |
| |
| int16_t AllocatePacketBufferByIdxSafe( |
| const WebRtcNetEQDecoder* used_codecs, |
| int16_t num_codecs, |
| const int16_t 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]; |
| |
| int16_t* neteq_packet_buffer_[MAX_NUM_SLAVE_NETEQ + 1]; |
| |
| int32_t 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]; |
| uint8_t num_slaves_; |
| bool received_stereo_; |
| void* master_slave_info_; |
| AudioFrame::VADActivity previous_audio_activity_; |
| |
| CriticalSectionWrapper* callback_crit_sect_; |
| // Minimum of "max number of packets," among all NetEq instances. |
| int min_of_max_num_packets_; |
| // Minimum of buffer-size among all NetEq instances. |
| int min_of_buffer_size_bytes_; |
| int per_packet_overhead_bytes_; |
| |
| // Keep track of AV-sync. Just used to set the slave when a slave is added. |
| bool av_sync_; |
| |
| int minimum_delay_ms_; |
| int maximum_delay_ms_; |
| }; |
| |
| } // namespace acm1 |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_ |
