
-Make ACM1 to depend on ACM2. -Remove APIs to set and get background noise mode. There is no VoE call to these APIs. -Remove APIs to set and get receive side VAD mode. There is no VoE call to these APIs, and NetEq 4, doesn't support them. -Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF detection. -Use acm_common_defs.h everywhere required. -Complete ACM factory method. -Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded. Remove dynamic payload-type assignment. BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2237004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4772 4adac7df-926f-26a2-2b94-8c16560cd09d
1224 lines
47 KiB
C++
1224 lines
47 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
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#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#define MAX_FRAME_SIZE_10MSEC 6
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// forward declaration
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struct WebRtcVadInst;
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struct WebRtcCngEncInst;
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namespace webrtc {
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// forward declaration
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struct CodecInst;
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namespace acm1 {
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class ACMNetEQ;
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class ACMGenericCodec {
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public:
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///////////////////////////////////////////////////////////////////////////
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// Constructor of the class
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//
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ACMGenericCodec();
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///////////////////////////////////////////////////////////////////////////
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// Destructor of the class.
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//
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virtual ~ACMGenericCodec();
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///////////////////////////////////////////////////////////////////////////
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// ACMGenericCodec* CreateInstance();
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// The function will be used for FEC. It is not implemented yet.
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//
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virtual ACMGenericCodec* CreateInstance() = 0;
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///////////////////////////////////////////////////////////////////////////
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// int16_t Encode()
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// The function is called to perform an encoding of the audio stored in
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// audio buffer. An encoding is performed only if enough audio, i.e. equal
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// to the frame-size of the codec, exist. The audio frame will be processed
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// by VAD and CN/DTX if required. There are few different cases.
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//
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// A) Neither VAD nor DTX is active; the frame is encoded by the encoder.
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//
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// B) VAD is enabled but not DTX; in this case the audio is processed by VAD
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// and encoded by the encoder. The "*encoding_type" will be either
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// "kActiveNormalEncode" or "kPassiveNormalEncode" if frame is active or
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// passive, respectively.
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//
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// C) DTX is enabled; if the codec has internal VAD/DTX we just encode the
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// frame by the encoder. Otherwise, the frame is passed through VAD and
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// if identified as passive, then it will be processed by CN/DTX. If the
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// frame is active it will be encoded by the encoder.
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//
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// This function acquires the appropriate locks and calls EncodeSafe() for
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// the actual processing.
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//
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// Outputs:
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// -bitstream : a buffer where bit-stream will be written to.
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// -bitstream_len_byte : contains the length of the bit-stream in
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// bytes.
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// -timestamp : contains the RTP timestamp, this is the
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// sampling time of the first sample encoded
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// (measured in number of samples).
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// -encoding_type : contains the type of encoding applied on the
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// audio samples. The alternatives are
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// (c.f. acm_common_types.h)
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// -kNoEncoding:
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// there was not enough data to encode. or
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// some error has happened that we could
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// not do encoding.
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// -kActiveNormalEncoded:
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// the audio frame is active and encoded by
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// the given codec.
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// -kPassiveNormalEncoded:
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// the audio frame is passive but coded with
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// the given codec (NO DTX).
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// -kPassiveDTXWB:
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// The audio frame is passive and used
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// wide-band CN to encode.
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// -kPassiveDTXNB:
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// The audio frame is passive and used
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// narrow-band CN to encode.
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//
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// Return value:
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// -1 if error is occurred, otherwise the length of the bit-stream in
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// bytes.
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//
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int16_t Encode(uint8_t* bitstream,
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int16_t* bitstream_len_byte,
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uint32_t* timestamp,
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WebRtcACMEncodingType* encoding_type);
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///////////////////////////////////////////////////////////////////////////
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// int16_t Decode()
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// This function is used to decode a given bit-stream, without engaging
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// NetEQ.
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//
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// This function acquires the appropriate locks and calls DecodeSafe() for
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// the actual processing. Please note that this is not functional yet.
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//
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// Inputs:
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// -bitstream : a buffer where bit-stream will be read.
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// -bitstream_len_byte : the length of the bit-stream in bytes.
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//
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// Outputs:
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// -audio : pointer to a buffer where the audio will written.
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// -audio_samples : number of audio samples out of decoding the given
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// bit-stream.
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// -speech_type : speech type (for future use).
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//
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// Return value:
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// -1 if failed to decode,
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// 0 if succeeded.
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//
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int16_t Decode(uint8_t* bitstream,
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int16_t bitstream_len_byte,
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int16_t* audio,
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int16_t* audio_samples,
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int8_t* speech_type);
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///////////////////////////////////////////////////////////////////////////
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// void SplitStereoPacket()
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// This function is used to split stereo payloads in left and right channel.
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// Codecs which has stereo support has there own implementation of the
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// function.
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//
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// Input/Output:
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// -payload : a vector with the received payload data.
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// The function will reorder the data so that
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// first half holds the left channel data, and the
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// second half the right channel data.
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// -payload_length : length of payload in bytes. Will be changed to
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// twice the input in case of true stereo, where
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// we simply copy the data and return it both for
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// left channel and right channel decoding.
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//
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virtual void SplitStereoPacket(uint8_t* /* payload */,
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int32_t* /* payload_length */) {}
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///////////////////////////////////////////////////////////////////////////
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// bool EncoderInitialized();
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//
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// Return value:
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// True if the encoder is successfully initialized,
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// false otherwise.
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//
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bool EncoderInitialized();
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///////////////////////////////////////////////////////////////////////////
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// bool DecoderInitialized();
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//
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// Return value:
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// True if the decoder is successfully initialized,
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// false otherwise.
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//
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bool DecoderInitialized();
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///////////////////////////////////////////////////////////////////////////
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// int16_t EncoderParams()
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// It is called to get encoder parameters. It will call
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// EncoderParamsSafe() in turn.
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//
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// Output:
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// -enc_params : a buffer where the encoder parameters is
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// written to. If the encoder is not
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// initialized this buffer is filled with
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// invalid values
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// Return value:
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// -1 if the encoder is not initialized,
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// 0 otherwise.
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//
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int16_t EncoderParams(WebRtcACMCodecParams *enc_params);
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///////////////////////////////////////////////////////////////////////////
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// int16_t DecoderParams(...)
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// It is called to get decoder parameters. It will call DecoderParamsSafe()
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// in turn.
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//
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// Output:
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// -dec_params : a buffer where the decoder parameters is
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// written to. If the decoder is not initialized
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// this buffer is filled with invalid values
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//
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// Return value:
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// -1 if the decoder is not initialized,
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// 0 otherwise.
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//
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//
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bool DecoderParams(WebRtcACMCodecParams *dec_params,
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const uint8_t payload_type);
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///////////////////////////////////////////////////////////////////////////
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// int16_t InitEncoder(...)
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// This function is called to initialize the encoder with the given
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// parameters.
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//
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// Input:
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// -codec_params : parameters of encoder.
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// -force_initialization: if false the initialization is invoked only if
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// the encoder is not initialized. If true the
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// encoder is forced to (re)initialize.
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//
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// Return value:
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// 0 if could initialize successfully,
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// -1 if failed to initialize.
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//
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//
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int16_t InitEncoder(WebRtcACMCodecParams* codec_params,
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bool force_initialization);
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///////////////////////////////////////////////////////////////////////////
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// int16_t InitDecoder()
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// This function is called to initialize the decoder with the given
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// parameters. (c.f. acm_common_defs.h & common_types.h for the
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// definition of the structure)
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//
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// Input:
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// -codec_params : parameters of decoder.
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// -force_initialization: if false the initialization is invoked only
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// if the decoder is not initialized. If true
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// the encoder is forced to(re)initialize.
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//
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// Return value:
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// 0 if could initialize successfully,
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// -1 if failed to initialize.
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//
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//
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int16_t InitDecoder(WebRtcACMCodecParams* codec_params,
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bool force_initialization);
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///////////////////////////////////////////////////////////////////////////
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// int32_t RegisterInNetEq(...)
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// This function is called to register the decoder in NetEq, with the given
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// payload type.
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//
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// Inputs:
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// -neteq : pointer to NetEq Instance
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// -codec_inst : instance with of the codec settings of the codec
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//
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// Return values
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// -1 if failed to register,
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// 0 if successfully initialized.
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//
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int32_t RegisterInNetEq(ACMNetEQ* neteq, const CodecInst& codec_inst);
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///////////////////////////////////////////////////////////////////////////
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// int32_t Add10MsData(...)
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// This function is called to add 10 ms of audio to the audio buffer of
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// the codec.
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//
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// Inputs:
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// -timestamp : the timestamp of the 10 ms audio. the timestamp
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// is the sampling time of the
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// first sample measured in number of samples.
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// -data : a buffer that contains the audio. The codec
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// expects to get the audio in correct sampling
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// frequency
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// -length : the length of the audio buffer
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// -audio_channel : 0 for mono, 1 for stereo (not supported yet)
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//
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// Return values:
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// -1 if failed
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// 0 otherwise.
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//
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int32_t Add10MsData(const uint32_t timestamp,
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const int16_t* data,
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const uint16_t length,
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const uint8_t audio_channel);
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///////////////////////////////////////////////////////////////////////////
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// uint32_t NoMissedSamples()
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// This function returns the number of samples which are overwritten in
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// the audio buffer. The audio samples are overwritten if the input audio
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// buffer is full, but Add10MsData() is called. (We might remove this
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// function if it is not used)
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//
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// Return Value:
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// Number of samples which are overwritten.
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//
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uint32_t NoMissedSamples() const;
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///////////////////////////////////////////////////////////////////////////
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// void ResetNoMissedSamples()
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// This function resets the number of overwritten samples to zero.
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// (We might remove this function if we remove NoMissedSamples())
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//
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void ResetNoMissedSamples();
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///////////////////////////////////////////////////////////////////////////
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// int16_t SetBitRate()
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// The function is called to set the encoding rate.
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//
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// Input:
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// -bitrate_bps : encoding rate in bits per second
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//
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// Return value:
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// -1 if failed to set the rate, due to invalid input or given
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// codec is not rate-adjustable.
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// 0 if the rate is adjusted successfully
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//
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int16_t SetBitRate(const int32_t bitrate_bps);
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///////////////////////////////////////////////////////////////////////////
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// DestructEncoderInst()
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// This API is used in conferencing. It will free the memory that is pointed
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// by |ptr_inst|. |ptr_inst| is a pointer to encoder instance, created and
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// filled up by calling EncoderInst(...).
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//
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// Inputs:
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// -ptr_inst : pointer to an encoder instance to be deleted.
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//
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//
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void DestructEncoderInst(void* ptr_inst);
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///////////////////////////////////////////////////////////////////////////
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// int16_t AudioBuffer()
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// This is used when synchronization of codecs is required. There are cases
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// that the audio buffers of two codecs have to be synched. By calling this
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// function on can get the audio buffer and other related parameters, such
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// as timestamps...
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//
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// Output:
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// -audio_buff : a pointer to WebRtcACMAudioBuff where the audio
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// buffer of this codec will be written to.
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//
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// Return value:
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// -1 if fails to copy the audio buffer,
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// 0 if succeeded.
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//
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int16_t AudioBuffer(WebRtcACMAudioBuff& audio_buff);
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///////////////////////////////////////////////////////////////////////////
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// uint32_t EarliestTimestamp()
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// Returns the timestamp of the first 10 ms in audio buffer. This is used
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// to identify if a synchronization of two encoders is required.
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//
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// Return value:
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// timestamp of the first 10 ms audio in the audio buffer.
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//
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uint32_t EarliestTimestamp() const;
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///////////////////////////////////////////////////////////////////////////
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// int16_t SetAudioBuffer()
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// This function is called to set the audio buffer and the associated
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// parameters to a given value.
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//
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// Return value:
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// -1 if fails to copy the audio buffer,
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// 0 if succeeded.
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//
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int16_t SetAudioBuffer(WebRtcACMAudioBuff& audio_buff);
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///////////////////////////////////////////////////////////////////////////
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// int16_t SetVAD()
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// This is called to set VAD & DTX. If the codec has internal DTX, it will
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// be used. If DTX is enabled and the codec does not have internal DTX,
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// WebRtc-VAD will be used to decide if the frame is active. If DTX is
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// disabled but VAD is enabled the audio is passed through VAD to label it
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// as active or passive, but the frame is encoded normally. However the
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// bit-stream is labeled properly so that ACM::Process() can use this
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// information. In case of failure, the previous states of the VAD & DTX
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// are kept.
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//
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// Input/Output:
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// -enable_dtx : if true DTX will be enabled otherwise the DTX is
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// disabled. If codec has internal DTX that will be
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// used, otherwise WebRtc-CNG is used. In the latter
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// case VAD is automatically activated.
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// -enable_vad : if true WebRtc-VAD is enabled, otherwise VAD is
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// disabled, except for the case that DTX is enabled
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// but codec doesn't have internal DTX. In this case
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// VAD is enabled regardless of the value of
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// |enable_vad|.
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// -mode : this specifies the aggressiveness of VAD.
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//
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// Return value
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// -1 if failed to set DTX & VAD as specified,
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// 0 if succeeded.
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//
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int16_t SetVAD(bool* enable_dtx,
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bool* enable_vad,
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ACMVADMode* mode);
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///////////////////////////////////////////////////////////////////////////
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// int32_t ReplaceInternalDTX()
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// This is called to replace the codec internal DTX with WebRtc DTX.
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// This is only valid for G729 where the user has possibility to replace
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// AnnexB with WebRtc DTX. For other codecs this function has no effect.
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//
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// Input:
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// -replace_internal_dtx : if true the internal DTX is replaced with WebRtc.
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//
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// Return value
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// -1 if failed to replace internal DTX,
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// 0 if succeeded.
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//
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int32_t ReplaceInternalDTX(const bool replace_internal_dtx);
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///////////////////////////////////////////////////////////////////////////
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// int32_t IsInternalDTXReplaced()
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// This is called to check if the codec internal DTX is replaced by WebRtc
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// DTX. This is only valid for G729 where the user has possibility to replace
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// AnnexB with WebRtc DTX. For other codecs this function has no effect.
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//
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// Output:
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// -internal_dtx_replaced: if true the internal DTX is replaced with WebRtc.
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//
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// Return value
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// -1 if failed to check
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// 0 if succeeded.
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//
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int32_t IsInternalDTXReplaced(bool* internal_dtx_replaced);
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///////////////////////////////////////////////////////////////////////////
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// void SetNetEqDecodeLock()
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// Passes the NetEq lock to the codec.
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//
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// Input:
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// -neteq_decode_lock : pointer to the lock associated with NetEQ of ACM.
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//
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void SetNetEqDecodeLock(RWLockWrapper* neteq_decode_lock) {
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neteq_decode_lock_ = neteq_decode_lock;
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}
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///////////////////////////////////////////////////////////////////////////
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// bool HasInternalDTX()
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// Used to check if the codec has internal DTX.
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//
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// Return value:
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// true if the codec has an internal DTX, e.g. G729,
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// false otherwise.
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//
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bool HasInternalDTX() const {
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return has_internal_dtx_;
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}
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///////////////////////////////////////////////////////////////////////////
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// int32_t GetEstimatedBandwidth()
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// Used to get decoder estimated bandwidth. Only iSAC will provide a value.
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//
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//
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// Return value:
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// -1 if fails to get decoder estimated bandwidth,
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// >0 estimated bandwidth in bits/sec.
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//
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int32_t GetEstimatedBandwidth();
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///////////////////////////////////////////////////////////////////////////
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// int32_t SetEstimatedBandwidth()
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// Used to set estiamted bandwidth sent out of band from other side. Only
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// iSAC will have use for the value.
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//
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// Input:
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// -estimated_bandwidth: estimated bandwidth in bits/sec
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//
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// Return value:
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// -1 if fails to set estimated bandwidth,
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// 0 on success.
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//
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int32_t SetEstimatedBandwidth(int32_t estimated_bandwidth);
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///////////////////////////////////////////////////////////////////////////
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// int32_t GetRedPayload()
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// Used to get codec specific RED payload (if such is implemented).
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// Currently only done in iSAC.
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//
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// Outputs:
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// -red_payload : a pointer to the data for RED payload.
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// -payload_bytes : number of bytes in RED payload.
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//
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// Return value:
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// -1 if fails to get codec specific RED,
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// 0 if succeeded.
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//
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int32_t GetRedPayload(uint8_t* red_payload,
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int16_t* payload_bytes);
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|
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///////////////////////////////////////////////////////////////////////////
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// int16_t ResetEncoder()
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// By calling this function you would re-initialize the encoder with the
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// current parameters. All the settings, e.g. VAD/DTX, frame-size... should
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// remain unchanged. (In case of iSAC we don't want to lose BWE history.)
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//
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// Return value
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// -1 if failed,
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// 0 if succeeded.
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//
|
|
int16_t ResetEncoder();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t ResetEncoder()
|
|
// By calling this function you would re-initialize the decoder with the
|
|
// current parameters.
|
|
//
|
|
// Return value
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
int16_t ResetDecoder(int16_t payload_type);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// void DestructEncoder()
|
|
// This function is called to delete the encoder instance, if possible, to
|
|
// have a fresh start. For codecs where encoder and decoder share the same
|
|
// instance we cannot delete the encoder and instead we will initialize the
|
|
// encoder. We also delete VAD and DTX if they have been created.
|
|
//
|
|
void DestructEncoder();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// void DestructDecoder()
|
|
// This function is called to delete the decoder instance, if possible, to
|
|
// have a fresh start. For codecs where encoder and decoder share the same
|
|
// instance we cannot delete the encoder and instead we will initialize the
|
|
// decoder. Before deleting decoder instance it has to be removed from the
|
|
// NetEq list.
|
|
//
|
|
void DestructDecoder();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t SamplesLeftToEncode()
|
|
// Returns the number of samples required to be able to do encoding.
|
|
//
|
|
// Return value:
|
|
// Number of samples.
|
|
//
|
|
int16_t SamplesLeftToEncode();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// uint32_t LastEncodedTimestamp()
|
|
// Returns the timestamp of the last frame it encoded.
|
|
//
|
|
// Return value:
|
|
// Timestamp.
|
|
//
|
|
uint32_t LastEncodedTimestamp() const;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// SetUniqueID()
|
|
// Set a unique ID for the codec to be used for tracing and debugging
|
|
//
|
|
// Input
|
|
// -id : A number to identify the codec.
|
|
//
|
|
void SetUniqueID(const uint32_t id);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// IsAudioBufferFresh()
|
|
// Specifies if ever audio is injected to this codec.
|
|
//
|
|
// Return value
|
|
// -true; no audio is feed into this codec
|
|
// -false; audio has already been fed to the codec.
|
|
//
|
|
bool IsAudioBufferFresh() const;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// UpdateDecoderSampFreq()
|
|
// For most of the codecs this function does nothing. It must be
|
|
// implemented for those codecs that one codec instance serves as the
|
|
// decoder for different flavors of the codec. One example is iSAC. there,
|
|
// iSAC 16 kHz and iSAC 32 kHz are treated as two different codecs with
|
|
// different payload types, however, there is only one iSAC instance to
|
|
// decode. The reason for that is we would like to decode and encode with
|
|
// the same codec instance for bandwidth estimator to work.
|
|
//
|
|
// Each time that we receive a new payload type, we call this function to
|
|
// prepare the decoder associated with the new payload. Normally, decoders
|
|
// doesn't have to do anything. For iSAC the decoder has to change it's
|
|
// sampling rate. The input parameter specifies the current flavor of the
|
|
// codec in codec database. For instance, if we just got a SWB payload then
|
|
// the input parameter is ACMCodecDB::isacswb.
|
|
//
|
|
// Input:
|
|
// -codec_id : the ID of the codec associated with the
|
|
// payload type that we just received.
|
|
//
|
|
// Return value:
|
|
// 0 if succeeded in updating the decoder.
|
|
// -1 if failed to update.
|
|
//
|
|
virtual int16_t UpdateDecoderSampFreq(int16_t /* codec_id */);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// UpdateEncoderSampFreq()
|
|
// Call this function to update the encoder sampling frequency. This
|
|
// is for codecs where one payload-name supports several encoder sampling
|
|
// frequencies. Otherwise, to change the sampling frequency we need to
|
|
// register new codec. ACM will consider that as registration of a new
|
|
// codec, not a change in parameter. For iSAC, switching from WB to SWB
|
|
// is treated as a change in parameter. Therefore, we need this function.
|
|
//
|
|
// Input:
|
|
// -samp_freq_hz : encoder sampling frequency.
|
|
//
|
|
// Return value:
|
|
// -1 if failed, or if this is meaningless for the given codec.
|
|
// 0 if succeeded.
|
|
//
|
|
virtual int16_t UpdateEncoderSampFreq(
|
|
uint16_t samp_freq_hz);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// EncoderSampFreq()
|
|
// Get the sampling frequency that the encoder (WebRtc wrapper) expects.
|
|
//
|
|
// Output:
|
|
// -samp_freq_hz : sampling frequency, in Hertz, which the encoder
|
|
// should be fed with.
|
|
//
|
|
// Return value:
|
|
// -1 if failed to output sampling rate.
|
|
// 0 if the sample rate is returned successfully.
|
|
//
|
|
virtual int16_t EncoderSampFreq(uint16_t& samp_freq_hz);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int32_t ConfigISACBandwidthEstimator()
|
|
// Call this function to configure the bandwidth estimator of ISAC.
|
|
// During the adaptation of bit-rate, iSAC automatically adjusts the
|
|
// frame-size (either 30 or 60 ms) to save on RTP header. The initial
|
|
// frame-size can be specified by the first argument. The configuration also
|
|
// regards the initial estimate of bandwidths. The estimator starts from
|
|
// this point and converges to the actual bottleneck. This is given by the
|
|
// second parameter. Furthermore, it is also possible to control the
|
|
// adaptation of frame-size. This is specified by the last parameter.
|
|
//
|
|
// Input:
|
|
// -init_frame_fize_ms : initial frame-size in milliseconds. For iSAC-wb
|
|
// 30 ms and 60 ms (default) are acceptable values,
|
|
// and for iSAC-swb 30 ms is the only acceptable
|
|
// value. Zero indicates default value.
|
|
// -init_rate_bps : initial estimate of the bandwidth. Values
|
|
// between 10000 and 58000 are acceptable.
|
|
// -enforce_frame_size : if true, the frame-size will not be adapted.
|
|
//
|
|
// Return value:
|
|
// -1 if failed to configure the bandwidth estimator,
|
|
// 0 if the configuration was successfully applied.
|
|
//
|
|
virtual int32_t ConfigISACBandwidthEstimator(
|
|
const uint8_t init_frame_size_msec,
|
|
const uint16_t init_rate_bps,
|
|
const bool enforce_frame_size);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// SetISACMaxPayloadSize()
|
|
// Set the maximum payload size of iSAC packets. No iSAC payload,
|
|
// regardless of its frame-size, may exceed the given limit. For
|
|
// an iSAC payload of size B bits and frame-size T sec we have;
|
|
// (B < max_payload_len_bytes * 8) and (B/T < max_rate_bit_per_sec), c.f.
|
|
// SetISACMaxRate().
|
|
//
|
|
// Input:
|
|
// -max_payload_len_bytes : maximum payload size in bytes.
|
|
//
|
|
// Return value:
|
|
// -1 if failed to set the maximum payload-size.
|
|
// 0 if the given length is set successfully.
|
|
//
|
|
virtual int32_t SetISACMaxPayloadSize(
|
|
const uint16_t max_payload_len_bytes);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// SetISACMaxRate()
|
|
// Set the maximum instantaneous rate of iSAC. For a payload of B bits
|
|
// with a frame-size of T sec the instantaneous rate is B/T bits per
|
|
// second. Therefore, (B/T < max_rate_bit_per_sec) and
|
|
// (B < max_payload_len_bytes * 8) are always satisfied for iSAC payloads,
|
|
// c.f SetISACMaxPayloadSize().
|
|
//
|
|
// Input:
|
|
// -max_rate_bps : maximum instantaneous bit-rate given in bits/sec.
|
|
//
|
|
// Return value:
|
|
// -1 if failed to set the maximum rate.
|
|
// 0 if the maximum rate is set successfully.
|
|
//
|
|
virtual int32_t SetISACMaxRate(const uint32_t max_rate_bps);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// SaveDecoderParamS()
|
|
// Save the parameters of decoder.
|
|
//
|
|
// Input:
|
|
// -codec_params : pointer to a structure where the parameters of
|
|
// decoder is stored in.
|
|
//
|
|
void SaveDecoderParam(const WebRtcACMCodecParams* codec_params);
|
|
|
|
int32_t FrameSize() {
|
|
return frame_len_smpl_;
|
|
}
|
|
|
|
void SetIsMaster(bool is_master);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// REDPayloadISAC()
|
|
// This is an iSAC-specific function. The function is called to get RED
|
|
// payload from a default-encoder.
|
|
//
|
|
// Inputs:
|
|
// -isac_rate : the target rate of the main payload. A RED
|
|
// payload is generated according to the rate of
|
|
// main payload. Note that we are not specifying the
|
|
// rate of RED payload, but the main payload.
|
|
// -isac_bw_estimate : bandwidth information should be inserted in
|
|
// RED payload.
|
|
//
|
|
// Output:
|
|
// -payload : pointer to a buffer where the RED payload will
|
|
// written to.
|
|
// -payload_len_bytes : a place-holder to write the length of the RED
|
|
// payload in Bytes.
|
|
//
|
|
// Return value:
|
|
// -1 if an error occurs, otherwise the length of the payload (in Bytes)
|
|
// is returned.
|
|
//
|
|
virtual int16_t REDPayloadISAC(const int32_t isac_rate,
|
|
const int16_t isac_bw_estimate,
|
|
uint8_t* payload,
|
|
int16_t* payload_len_bytes);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// IsTrueStereoCodec()
|
|
// Call to see if current encoder is a true stereo codec. This function
|
|
// should be overwritten for codecs which are true stereo codecs
|
|
// Return value:
|
|
// -true if stereo codec
|
|
// -false if not stereo codec.
|
|
//
|
|
virtual bool IsTrueStereoCodec();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// HasFrameToEncode()
|
|
// Returns true if there is enough audio buffered for encoding, such that
|
|
// calling Encode() will return a payload.
|
|
//
|
|
bool HasFrameToEncode() const;
|
|
|
|
protected:
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// All the functions with FunctionNameSafe(...) contain the actual
|
|
// implementation of FunctionName(...). FunctionName() acquires an
|
|
// appropriate lock and calls FunctionNameSafe() to do the actual work.
|
|
// Therefore, for the description of functionality, input/output arguments
|
|
// and return value we refer to FunctionName()
|
|
//
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See Decode() for the description of function, input(s)/output(s) and
|
|
// return value.
|
|
//
|
|
virtual int16_t DecodeSafe(uint8_t* bitstream,
|
|
int16_t bitstream_len_byte,
|
|
int16_t* audio,
|
|
int16_t* audio_samples,
|
|
int8_t* speech_type) = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See Add10MsSafe() for the description of function, input(s)/output(s)
|
|
// and return value.
|
|
//
|
|
virtual int32_t Add10MsDataSafe(const uint32_t timestamp,
|
|
const int16_t* data,
|
|
const uint16_t length,
|
|
const uint8_t audio_channel);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See RegisterInNetEq() for the description of function,
|
|
// input(s)/output(s) and return value.
|
|
//
|
|
virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
|
|
const CodecInst& codec_inst) = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See EncoderParam() for the description of function, input(s)/output(s)
|
|
// and return value.
|
|
//
|
|
int16_t EncoderParamsSafe(WebRtcACMCodecParams *enc_params);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See DecoderParam for the description of function, input(s)/output(s)
|
|
// and return value.
|
|
//
|
|
// Note:
|
|
// Any Class where a single instance handle several flavors of the
|
|
// same codec, therefore, several payload types are associated with
|
|
// the same instance have to implement this function.
|
|
//
|
|
// Currently only iSAC is implementing it. A single iSAC instance is
|
|
// used for decoding both WB & SWB stream. At one moment both WB & SWB
|
|
// can be registered as receive codec. Hence two payloads are associated
|
|
// with a single codec instance.
|
|
//
|
|
virtual bool DecoderParamsSafe(WebRtcACMCodecParams *dec_params,
|
|
const uint8_t payload_type);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See ResetEncoder() for the description of function, input(s)/output(s)
|
|
// and return value.
|
|
//
|
|
int16_t ResetEncoderSafe();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See InitEncoder() for the description of function, input(s)/output(s)
|
|
// and return value.
|
|
//
|
|
int16_t InitEncoderSafe(WebRtcACMCodecParams *codec_params,
|
|
bool force_initialization);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See InitDecoder() for the description of function, input(s)/output(s)
|
|
// and return value.
|
|
//
|
|
int16_t InitDecoderSafe(WebRtcACMCodecParams *codec_params,
|
|
bool force_initialization);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See ResetDecoder() for the description of function, input(s)/output(s)
|
|
// and return value.
|
|
//
|
|
int16_t ResetDecoderSafe(int16_t payload_type);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See DestructEncoder() for the description of function,
|
|
// input(s)/output(s) and return value.
|
|
//
|
|
virtual void DestructEncoderSafe() = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See DestructDecoder() for the description of function,
|
|
// input(s)/output(s) and return value.
|
|
//
|
|
virtual void DestructDecoderSafe() = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See SetBitRate() for the description of function, input(s)/output(s)
|
|
// and return value.
|
|
//
|
|
// Any codec that can change the bit-rate has to implement this.
|
|
//
|
|
virtual int16_t SetBitRateSafe(const int32_t bitrate_bps);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See GetEstimatedBandwidth() for the description of function,
|
|
// input(s)/output(s) and return value.
|
|
//
|
|
virtual int32_t GetEstimatedBandwidthSafe();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See SetEstimatedBandwidth() for the description of function,
|
|
// input(s)/output(s) and return value.
|
|
//
|
|
virtual int32_t SetEstimatedBandwidthSafe(
|
|
int32_t estimated_bandwidth);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See GetRedPayload() for the description of function, input(s)/output(s)
|
|
// and return value.
|
|
//
|
|
virtual int32_t GetRedPayloadSafe(uint8_t* red_payload,
|
|
int16_t* payload_bytes);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See SetVAD() for the description of function, input(s)/output(s) and
|
|
// return value.
|
|
//
|
|
int16_t SetVADSafe(bool* enable_dtx,
|
|
bool* enable_vad,
|
|
ACMVADMode* mode);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See ReplaceInternalDTX() for the description of function, input and
|
|
// return value.
|
|
//
|
|
virtual int32_t ReplaceInternalDTXSafe(const bool replace_internal_dtx);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// See IsInternalDTXReplaced() for the description of function, input and
|
|
// return value.
|
|
//
|
|
virtual int32_t IsInternalDTXReplacedSafe(bool* internal_dtx_replaced);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t CreateEncoder()
|
|
// Creates the encoder instance.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
int16_t CreateEncoder();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t CreateDecoder()
|
|
// Creates the decoder instance.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
int16_t CreateDecoder();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t EnableVAD();
|
|
// Enables VAD with the given mode. The VAD instance will be created if
|
|
// it does not exists.
|
|
//
|
|
// Input:
|
|
// -mode : VAD mode c.f. audio_coding_module_typedefs.h for
|
|
// the options.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
int16_t EnableVAD(ACMVADMode mode);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t DisableVAD()
|
|
// Disables VAD.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
int16_t DisableVAD();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t EnableDTX()
|
|
// Enables DTX. This method should be overwritten for codecs which have
|
|
// internal DTX.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
virtual int16_t EnableDTX();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t DisableDTX()
|
|
// Disables usage of DTX. This method should be overwritten for codecs which
|
|
// have internal DTX.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
virtual int16_t DisableDTX();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t InternalEncode()
|
|
// This is a codec-specific function called in EncodeSafe() to actually
|
|
// encode a frame of audio.
|
|
//
|
|
// Outputs:
|
|
// -bitstream : pointer to a buffer where the bit-stream is
|
|
// written to.
|
|
// -bitstream_len_byte : the length of the bit-stream in bytes,
|
|
// a negative value indicates error.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// otherwise the length of the bit-stream is returned.
|
|
//
|
|
virtual int16_t InternalEncode(uint8_t* bitstream,
|
|
int16_t* bitstream_len_byte) = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t InternalInitEncoder()
|
|
// This is a codec-specific function called in InitEncoderSafe(), it has to
|
|
// do all codec-specific operation to initialize the encoder given the
|
|
// encoder parameters.
|
|
//
|
|
// Input:
|
|
// -codec_params : pointer to a structure that contains parameters to
|
|
// initialize encoder.
|
|
// Set codec_params->codec_inst.rate to -1 for
|
|
// iSAC to operate in adaptive mode.
|
|
// (to do: if frame-length is -1 frame-length will be
|
|
// automatically adjusted, otherwise, given
|
|
// frame-length is forced)
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
virtual int16_t InternalInitEncoder(
|
|
WebRtcACMCodecParams *codec_params) = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t InternalInitDecoder()
|
|
// This is a codec-specific function called in InitDecoderSafe(), it has to
|
|
// do all codec-specific operation to initialize the decoder given the
|
|
// decoder parameters.
|
|
//
|
|
// Input:
|
|
// -codec_params : pointer to a structure that contains parameters to
|
|
// initialize encoder.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
virtual int16_t InternalInitDecoder(
|
|
WebRtcACMCodecParams *codec_params) = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// void IncreaseNoMissedSamples()
|
|
// This method is called to increase the number of samples that are
|
|
// overwritten in the audio buffer.
|
|
//
|
|
// Input:
|
|
// -num_samples : the number of overwritten samples is incremented
|
|
// by this value.
|
|
//
|
|
void IncreaseNoMissedSamples(const int16_t num_samples);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t InternalCreateEncoder()
|
|
// This is a codec-specific method called in CreateEncoderSafe() it is
|
|
// supposed to perform all codec-specific operations to create encoder
|
|
// instance.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
virtual int16_t InternalCreateEncoder() = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t InternalCreateDecoder()
|
|
// This is a codec-specific method called in CreateDecoderSafe() it is
|
|
// supposed to perform all codec-specific operations to create decoder
|
|
// instance.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
virtual int16_t InternalCreateDecoder() = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// void InternalDestructEncoderInst()
|
|
// This is a codec-specific method, used in conferencing, called from
|
|
// DestructEncoderInst(). The input argument is pointer to encoder instance
|
|
// (codec instance for codecs that encoder and decoder share the same
|
|
// instance). This method is called to free the memory that |ptr_inst| is
|
|
// pointing to.
|
|
//
|
|
// Input:
|
|
// -ptr_inst : pointer to encoder instance.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
virtual void InternalDestructEncoderInst(void* ptr_inst) = 0;
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t InternalResetEncoder()
|
|
// This method is called to reset the states of encoder. However, the
|
|
// current parameters, e.g. frame-length, should remain as they are. For
|
|
// most of the codecs a re-initialization of the encoder is what needs to
|
|
// be down. But for iSAC we like to keep the BWE history so we cannot
|
|
// re-initialize. As soon as such an API is implemented in iSAC this method
|
|
// has to be overwritten in ACMISAC class.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
virtual int16_t InternalResetEncoder();
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// int16_t ProcessFrameVADDTX()
|
|
// This function is called when a full frame of audio is available. It will
|
|
// break the audio frame into blocks such that each block could be processed
|
|
// by VAD & CN/DTX. If a frame is divided into two blocks then there are two
|
|
// cases. First, the first block is active, the second block will not be
|
|
// processed by CN/DTX but only by VAD and return to caller with
|
|
// '*samples_processed' set to zero. There, the audio frame will be encoded
|
|
// by the encoder. Second, the first block is inactive and is processed by
|
|
// CN/DTX, then we stop processing the next block and return to the caller
|
|
// which is EncodeSafe(), with "*samples_processed" equal to the number of
|
|
// samples in first block.
|
|
//
|
|
// Output:
|
|
// -bitstream : pointer to a buffer where DTX frame, if
|
|
// generated, will be written to.
|
|
// -bitstream_len_byte : contains the length of bit-stream in bytes, if
|
|
// generated. Zero if no bit-stream is generated.
|
|
// -samples_processed : contains no of samples that actually CN has
|
|
// processed. Those samples processed by CN will not
|
|
// be encoded by the encoder, obviously. If
|
|
// contains zero, it means that the frame has been
|
|
// identified as active by VAD. Note that
|
|
// "*samples_processed" might be non-zero but
|
|
// "*bitstream_len_byte" be zero.
|
|
//
|
|
// Return value:
|
|
// -1 if failed,
|
|
// 0 if succeeded.
|
|
//
|
|
int16_t ProcessFrameVADDTX(uint8_t* bitstream,
|
|
int16_t* bitstream_len_byte,
|
|
int16_t* samples_processed);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// CanChangeEncodingParam()
|
|
// Check if the codec parameters can be changed. In conferencing normally
|
|
// codec parameters cannot be changed. The exception is bit-rate of isac.
|
|
//
|
|
// return value:
|
|
// -true if codec parameters are allowed to change.
|
|
// -false otherwise.
|
|
//
|
|
virtual bool CanChangeEncodingParam(CodecInst& codec_inst);
|
|
|
|
///////////////////////////////////////////////////////////////////////////
|
|
// CurrentRate()
|
|
// Call to get the current encoding rate of the encoder. This function
|
|
// should be overwritten for codecs which automatically change their
|
|
// target rate. One example is iSAC. The output of the function is the
|
|
// current target rate.
|
|
//
|
|
// Output:
|
|
// -rate_bps : the current target rate of the codec.
|
|
//
|
|
virtual void CurrentRate(int32_t& /* rate_bps */);
|
|
|
|
virtual void SaveDecoderParamSafe(const WebRtcACMCodecParams* codec_params);
|
|
|
|
// &in_audio_[in_audio_ix_write_] always point to where new audio can be
|
|
// written to
|
|
int16_t in_audio_ix_write_;
|
|
|
|
// &in_audio_[in_audio_ix_read_] points to where audio has to be read from
|
|
int16_t in_audio_ix_read_;
|
|
|
|
int16_t in_timestamp_ix_write_;
|
|
|
|
// Where the audio is stored before encoding,
|
|
// To save memory the following buffer can be allocated
|
|
// dynamically for 80 ms depending on the sampling frequency
|
|
// of the codec.
|
|
int16_t* in_audio_;
|
|
uint32_t* in_timestamp_;
|
|
|
|
int16_t frame_len_smpl_;
|
|
uint16_t num_channels_;
|
|
|
|
// This will point to a static database of the supported codecs
|
|
int16_t codec_id_;
|
|
|
|
// This will account for the number of samples were not encoded
|
|
// the case is rare, either samples are missed due to overwrite
|
|
// at input buffer or due to encoding error
|
|
uint32_t num_missed_samples_;
|
|
|
|
// True if the encoder instance created
|
|
bool encoder_exist_;
|
|
bool decoder_exist_;
|
|
// True if the encoder instance initialized
|
|
bool encoder_initialized_;
|
|
bool decoder_initialized_;
|
|
|
|
bool registered_in_neteq_;
|
|
|
|
// VAD/DTX
|
|
bool has_internal_dtx_;
|
|
WebRtcVadInst* ptr_vad_inst_;
|
|
bool vad_enabled_;
|
|
ACMVADMode vad_mode_;
|
|
int16_t vad_label_[MAX_FRAME_SIZE_10MSEC];
|
|
bool dtx_enabled_;
|
|
WebRtcCngEncInst* ptr_dtx_inst_;
|
|
uint8_t num_lpc_params_;
|
|
bool sent_cn_previous_;
|
|
bool is_master_;
|
|
int16_t prev_frame_cng_;
|
|
|
|
WebRtcACMCodecParams encoder_params_;
|
|
WebRtcACMCodecParams decoder_params_;
|
|
|
|
// Used as a global lock for all available decoders
|
|
// so that no decoder is used when NetEQ decodes.
|
|
RWLockWrapper* neteq_decode_lock_;
|
|
// Used to lock wrapper internal data
|
|
// such as buffers and state variables.
|
|
RWLockWrapper& codec_wrapper_lock_;
|
|
|
|
uint32_t last_encoded_timestamp_;
|
|
uint32_t last_timestamp_;
|
|
bool is_audio_buff_fresh_;
|
|
uint32_t unique_id_;
|
|
};
|
|
|
|
} // namespace acm1
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
|