--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/spandsp-0.0.3/spandsp-0.0.3/src/spandsp/echo.h	Fri Jun 25 16:00:21 2010 +0200
@@ -0,0 +1,220 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * echo.c - A line echo canceller.  This code is being developed
+ *          against and partially complies with G168.
+ *
+ * Written by Steve Underwood <steveu@coppice.org> 
+ *         and David Rowe <david_at_rowetel_dot_com>
+ *
+ * Copyright (C) 2001 Steve Underwood and 2007 David Rowe
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: echo.h,v 1.9 2006/10/24 13:45:28 steveu Exp $
+ */
+
+/*! \file */
+
+#if !defined(_ECHO_H_)
+#define _ECHO_H_
+
+/*! \page echo_can_page Line echo cancellation for voice
+
+\section echo_can_page_sec_1 What does it do?
+This module aims to provide G.168-2002 compliant echo cancellation, to remove
+electrical echoes (e.g. from 2-4 wire hybrids) from voice calls.
+
+\section echo_can_page_sec_2 How does it work?
+The heart of the echo cancellor is FIR filter. This is adapted to match the echo
+impulse response of the telephone line. It must be long enough to adequately cover
+the duration of that impulse response. The signal transmitted to the telephone line
+is passed through the FIR filter. Once the FIR is properly adapted, the resulting
+output is an estimate of the echo signal received from the line. This is subtracted
+from the received signal. The result is an estimate of the signal which originated
+at the far end of the line, free from echos of our own transmitted signal. 
+
+The least mean squares (LMS) algorithm is attributed to Widrow and Hoff, and was
+introduced in 1960. It is the commonest form of filter adaption used in things
+like modem line equalisers and line echo cancellers. There it works very well.
+However, it only works well for signals of constant amplitude. It works very poorly
+for things like speech echo cancellation, where the signal level varies widely.
+This is quite easy to fix. If the signal level is normalised - similar to applying
+AGC - LMS can work as well for a signal of varying amplitude as it does for a modem
+signal. This normalised least mean squares (NLMS) algorithm is the commonest one used
+for speech echo cancellation. Many other algorithms exist - e.g. RLS (essentially
+the same as Kalman filtering), FAP, etc. Some perform significantly better than NLMS.
+However, factors such as computational complexity and patents favour the use of NLMS.
+
+A simple refinement to NLMS can improve its performance with speech. NLMS tends
+to adapt best to the strongest parts of a signal. If the signal is white noise,
+the NLMS algorithm works very well. However, speech has more low frequency than
+high frequency content. Pre-whitening (i.e. filtering the signal to flatten
+its spectrum) the echo signal improves the adapt rate for speech, and ensures the
+final residual signal is not heavily biased towards high frequencies. A very low
+complexity filter is adequate for this, so pre-whitening adds little to the
+compute requirements of the echo canceller.
+
+An FIR filter adapted using pre-whitened NLMS performs well, provided certain
+conditions are met: 
+
+    - The transmitted signal has poor self-correlation.
+    - There is no signal being generated within the environment being cancelled.
+
+The difficulty is that neither of these can be guaranteed.
+
+If the adaption is performed while transmitting noise (or something fairly noise
+like, such as voice) the adaption works very well. If the adaption is performed
+while transmitting something highly correlative (typically narrow band energy
+such as signalling tones or DTMF), the adaption can go seriously wrong. The reason
+is there is only one solution for the adaption on a near random signal - the impulse
+response of the line. For a repetitive signal, there are any number of solutions
+which converge the adaption, and nothing guides the adaption to choose the generalised
+one. Allowing an untrained canceller to converge on this kind of narrowband
+energy probably a good thing, since at least it cancels the tones. Allowing a well
+converged canceller to continue converging on such energy is just a way to ruin
+its generalised adaption. A narrowband detector is needed, so adapation can be
+suspended at appropriate times.
+
+The adaption process is based on trying to eliminate the received signal. When
+there is any signal from within the environment being cancelled it may upset the
+adaption process. Similarly, if the signal we are transmitting is small, noise
+may dominate and disturb the adaption process. If we can ensure that the
+adaption is only performed when we are transmitting a significant signal level,
+and the environment is not, things will be OK. Clearly, it is easy to tell when
+we are sending a significant signal. Telling, if the environment is generating a
+significant signal, and doing it with sufficient speed that the adaption will
+not have diverged too much more we stop it, is a little harder. 
+
+The key problem in detecting when the environment is sourcing significant energy
+is that we must do this very quickly. Given a reasonably long sample of the
+received signal, there are a number of strategies which may be used to assess
+whether that signal contains a strong far end component. However, by the time
+that assessment is complete the far end signal will have already caused major
+mis-convergence in the adaption process. An assessment algorithm is needed which
+produces a fairly accurate result from a very short burst of far end energy. 
+
+\section echo_can_page_sec_3 How do I use it?
+The echo cancellor processes both the transmit and receive streams sample by
+sample. The processing function is not declared inline. Unfortunately,
+cancellation requires many operations per sample, so the call overhead is only a
+minor burden. 
+*/
+
+#include "fir.h"
+
+/* Mask bits for the adaption mode */
+
+#define ECHO_CAN_USE_ADAPTION       0x01
+#define ECHO_CAN_USE_NLP            0x02
+#define ECHO_CAN_USE_CNG            0x04
+#define ECHO_CAN_USE_CLIP           0x08
+#define ECHO_CAN_USE_TX_HPF         0x10
+#define ECHO_CAN_USE_RX_HPF         0x20
+#define ECHO_CAN_DISABLE            0x40
+
+/*!
+    G.168 echo canceller descriptor. This defines the working state for a line
+    echo canceller.
+*/
+typedef struct
+{
+    int16_t tx,rx;
+    int16_t clean;
+    int16_t clean_nlp;
+
+    int nonupdate_dwell;
+    int curr_pos;	
+    int taps;
+    int log2taps;
+    int adaption_mode;
+
+    int cond_met;
+    int32_t Pstates;
+    int16_t adapt;
+    int32_t factor;
+    int16_t shift;
+
+    /* Average levels and averaging filter states */ 
+    int Ltxacc, Lrxacc, Lcleanacc, Lclean_bgacc;
+    int Ltx, Lrx;
+    int Lclean;
+    int Lclean_bg;
+    int Lbgn, Lbgn_acc, Lbgn_upper, Lbgn_upper_acc;
+
+    /* foreground and background filter states */
+    fir16_state_t fir_state;
+    fir16_state_t fir_state_bg;
+    int16_t *fir_taps16[2];
+    
+    /* DC blocking filter states */
+    int tx_1, tx_2, rx_1, rx_2;
+   
+    /* optional High Pass Filter states */
+    int32_t xvtx[5], yvtx[5];
+    int32_t xvrx[5], yvrx[5];
+   
+    /* Parameters for the optional Hoth noise generator */
+    int cng_level;
+    int cng_rndnum;
+    int cng_filter;
+    
+    /* snapshot sample of coeffs used for development */
+    int16_t *snapshot;       
+
+} echo_can_state_t;
+
+/*! Create a voice echo canceller context.
+    \param len The length of the canceller, in samples.
+    \return The new canceller context, or NULL if the canceller could not be created.
+*/
+echo_can_state_t *echo_can_create(int len, int adaption_mode);
+
+/*! Free a voice echo canceller context.
+    \param ec The echo canceller context.
+*/
+void echo_can_free(echo_can_state_t *ec);
+
+/*! Flush (reinitialise) a voice echo canceller context.
+    \param ec The echo canceller context.
+*/
+void echo_can_flush(echo_can_state_t *ec);
+
+/*! Set the adaption mode of a voice echo canceller context.
+    \param ec The echo canceller context.
+    \param adapt The mode.
+*/
+void echo_can_adaption_mode(echo_can_state_t *ec, int adaption_mode);
+
+void echo_can_snapshot(echo_can_state_t *ec);
+
+/*! Process a sample through a voice echo canceller.
+    \param ec The echo canceller context.
+    \param tx The transmitted audio sample.
+    \param rx The received audio sample.
+    \return The clean (echo cancelled) received sample.
+*/
+int16_t echo_can_update(echo_can_state_t *ec, int16_t tx, int16_t rx);
+
+/*! Process to high pass filter the tx signal.
+    \param ec The echo canceller context.
+    \param tx The transmitted auio sample.
+    \return The HP filtered transmit sample, send this to your D/A.
+*/
+int16_t echo_can_hpf_tx(echo_can_state_t *ec, int16_t tx);
+
+#endif
+/*- End of file ------------------------------------------------------------*/