diff --git a/src/audio/audiodevice.cpp b/src/audio/audiodevice.cpp
index 765fc6be9b1ab3f995370420552a1a48e108aca7..a4672da846bd13b1b2e369824ae9d007fc73690a 100644
--- a/src/audio/audiodevice.cpp
+++ b/src/audio/audiodevice.cpp
@@ -22,8 +22,8 @@
const double AudioDevice::DEFAULT_RATE = 8000.0;
-AudioDevice::AudioDevice(int id, const std::string& apiName, const std::string& name) :
- _id(id), _apiName(apiName), _name(name)
+AudioDevice::AudioDevice(int id, const std::string& name) :
+ _id(id), _name(name)
{
_rate = DEFAULT_RATE;
}
diff --git a/src/audio/audiodevice.h b/src/audio/audiodevice.h
index acdfee46368c8f9d187a6d82556600ad4ef8dabb..64c884be9a133748cb9155b251449a076937ba02 100644
--- a/src/audio/audiodevice.h
+++ b/src/audio/audiodevice.h
@@ -20,36 +20,62 @@
#ifndef _AUDIO_DEVICE_H
#define _AUDIO_DEVICE_H
-
#include <string>
#define AUDIODEVICERATE 8000
/**
- * @file audiodevice.c
+ * @file audiodevice.h
* @brief Container device for attribute storage
* Have almost only get/set method
*/
class AudioDevice {
public:
- AudioDevice(int id, const std::string& apiName, const std::string& name);
+ /**
+ * Constructor
+ * @param id
+ * @param
+ * @param
+ */
+ AudioDevice(int id, const std::string& name);
+
+ /**
+ * Destructor
+ */
~AudioDevice();
+ /** Default sample rate */
const static double DEFAULT_RATE;
+ /**
+ * Read accessor to the ID
+ * @return int The ID of the audiodevice
+ */
int getId() { return _id; }
- const std::string& getApiName() {return _apiName; }
+
+ /**
+ * Read accessor to the name
+ * @return std::string& A string description
+ */
const std::string& getName() {return _name; }
+ /**
+ * Write accessor to the sample rate
+ * @param rate The sample rate
+ */
void setRate(double rate) { _rate = rate;}
+
+ /**
+ * Read accessor to the sample rate
+ * @return double The sample rate
+ */
double getRate() { return _rate; }
private:
/** Integer id of the device, can not be 0 */
int _id;
- /** Host API Name, ex: OSS, ALSA */
- std::string _apiName;
+
/** Name of the device */
std::string _name;
diff --git a/src/audio/audiofile.h b/src/audio/audiofile.h
index d615e358f42477eaaebd1675a323ecc913811dc7..1de076de13f992ee77ebf2112a51fa5b0a3548be 100644
--- a/src/audio/audiofile.h
+++ b/src/audio/audiofile.h
@@ -28,22 +28,57 @@
#include "codecDescriptor.h"
/**
- @author Yan Morin <yan.morin@savoirfairelinux.com>
-*/
+ * @file audiofile.h
+ * @brief A class to manage sound files
+ */
+
class AudioFile : public AudioLoop
{
public:
+ /**
+ * Constructor
+ */
AudioFile();
+
+ /**
+ * Destructor
+ */
~AudioFile();
- bool loadFile(const std::string& filename, AudioCodec *codec , unsigned int sampleRate/*=8000*/);
+ /**
+ * Load a sound file in memory
+ * @param filename The absolute path to the file
+ * @param codec The codec to decode and encode it
+ * @param sampleRate The sample rate to read it
+ * @return bool True on success
+ */
+ bool loadFile(const std::string& filename, AudioCodec *codec , unsigned int sampleRate);
+
+ /**
+ * Start the sound file
+ */
void start() { _start = true; }
+
+ /**
+ * Stop the sound file
+ */
void stop() { _start = false; }
+
+ /**
+ * Tells whether or not the file is playing
+ * @return bool True if yes
+ * false otherwise
+ */
bool isStarted() { return _start; }
private:
+ /** The absolute path to the sound file */
std::string _filename;
+
+ /** Your preferred codec */
AudioCodec* _codec;
+
+ /** Start or not */
bool _start;
};
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index ff13e4f9c566082804bbfee7751873cd63f8c4fc..183f8231e26709b4687f12d9d5e0eb29284ebaad 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -1,9 +1,8 @@
/*
- * Copyright (C) 2005 Savoir-Faire Linux inc.
+ * Copyright (C) 2008 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Jerome Oufella <jerome.oufella@savoirfairelinux.com>
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Guillaume Carmel-Archambault <guillaume.carmel-archambault@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -37,7 +36,6 @@
AudioLayer::AudioLayer(ManagerImpl* manager)
: _defaultVolume(100)
- , _errorMessage(-1)
, _manager(manager)
, _PlaybackHandle( NULL )
, _CaptureHandle( NULL )
@@ -123,13 +121,6 @@ AudioLayer::stopStream(void)
_urgentBuffer.flush();
}
- int
-AudioLayer::getDeviceCount()
-{
- // TODO: everything
- return 1;
-}
-
void AudioLayer::AlsaCallBack( snd_async_handler_t* pcm_callback )
{
( ( AudioLayer *)snd_async_handler_get_callback_private( pcm_callback )) -> playTones();
@@ -326,8 +317,6 @@ AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
if( err = snd_pcm_hw_params_set_format( _CaptureHandle, hwParams, SND_PCM_FORMAT_S16_LE) < 0) _debugAlsa(" Cannot set sample format (%s)\n", snd_strerror(err));
if( err = snd_pcm_hw_params_set_rate_near( _CaptureHandle, hwParams, &rate_in, &dir) < 0) _debugAlsa(" Cannot set sample rate (%s)\n", snd_strerror(err));
if( err = snd_pcm_hw_params_set_channels( _CaptureHandle, hwParams, 1) < 0) _debugAlsa(" Cannot set channel count (%s)\n", snd_strerror(err));
- //if( err = snd_pcm_hw_params_set_period_size_near( _CaptureHandle, hwParams, &period_size_in , &dir) < 0) _debugAlsa(" Cannot set period size (%s)\n", snd_strerror(err));
- //if( err = snd_pcm_hw_params_set_buffer_size_near( _CaptureHandle, hwParams, &buffer_size_in ) < 0) _debugAlsa(" Cannot set buffer size (%s)\n", snd_strerror(err));
if( err = snd_pcm_hw_params_set_period_time_near( _CaptureHandle, hwParams, &period_time , &dir) < 0) _debugAlsa(" Cannot set period time (%s)\n", snd_strerror(err));
if( err = snd_pcm_hw_params_set_buffer_time_near( _CaptureHandle, hwParams, &buffer_time , &dir) < 0) _debugAlsa(" Cannot set buffer time (%s)\n", snd_strerror(err));
if( err = snd_pcm_hw_params_get_period_size( hwParams, &period_size_in , &dir) < 0) _debugAlsa(" Cannot get period size (%s)\n", snd_strerror(err));
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 18426aa57af21daf3008ee15b9f0bea541fdc8b6..e56f5ad7a48bf13f2c3adfc3b75d4e5ced95da58 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -1,9 +1,8 @@
/*
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+ * Copyright (C) 2004-2008 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Jerome Oufella <jerome.oufella@savoirfairelinux.com>
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Guillaume Carmel-Archambault <guillaume.carmel-archambault@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -23,29 +22,40 @@
#ifndef _AUDIO_LAYER_H
#define _AUDIO_LAYER_H
-#include <cc++/thread.h> // for ost::Mutex
-
#include "../global.h"
#include "audiodevice.h"
#include "ringbuffer.h"
+#include <cc++/thread.h> // for ost::Mutex
#include <vector>
#include <alsa/asoundlib.h>
#include <iostream>
-#include <fstream>
#include <istream>
#include <sstream>
#define FRAME_PER_BUFFER 160
-
class RingBuffer;
class ManagerImpl;
+/** Associate a sound card index to its string description */
typedef std::pair<int , std::string> HwIDPair;
+/**
+ * @file audiolayer.h
+ * @brief Main sound class. Manages the data transfers between the application and the hardware.
+ */
+
class AudioLayer {
public:
+ /**
+ * Constructor
+ * @param manager An instance of managerimpl
+ */
AudioLayer(ManagerImpl* manager);
+
+ /**
+ * Destructor
+ */
~AudioLayer(void);
/**
@@ -61,7 +71,7 @@ class AudioLayer {
* SFL_PCM_BOTH
* @param plugin The alsa plugin ( dmix , default , front , surround , ...)
*/
- bool openDevice(int, int, int, int, int, std::string);
+ bool openDevice(int indexIn, int indexOut, int sampleRate, int frameSize, int stream, std::string plugin);
/**
* Start the capture stream and prepare the playback stream.
@@ -107,8 +117,9 @@ class AudioLayer {
/**
* Send samples to the audio device.
- * @params buffer The buffer containing the data to be played ( voice and DTMF )
- * @params toCopy The number of samples, in bytes
+ * @param buffer The buffer containing the data to be played ( voice and DTMF )
+ * @param toCopy The number of samples, in bytes
+ * @param isTalking If whether or not the conversation is running
* @return int The number of bytes played
*/
int playSamples(void* buffer, int toCopy, bool isTalking);
@@ -154,22 +165,50 @@ class AudioLayer {
* @return std::vector<std::string> The vector containing the string description of the card
*/
std::vector<std::string> getSoundCardsInfo( int stream );
+
+ /**
+ * Check if the given index corresponds to an existing sound card and supports the specified streaming mode
+ * @param card An index
+ * @param stream The stream mode
+ * SFL_PCM_CAPTURE
+ * SFL_PCM_PLAYBACK
+ * SFL_PCM_BOTH
+ * @return bool True if it exists and supports the mode
+ * false otherwise
+ */
bool soundCardIndexExist( int card , int stream );
+
+ /**
+ * An index is associated with its string description
+ * @param description The string description
+ * @return int Its index
+ */
int soundCardGetIndex( std::string description );
+ /**
+ * Write accessor to the error state
+ * @param error The error code
+ * Could be: ALSA_PLAYBACK_DEVICE
+ * ALSA_CAPTURE_DEVICE
+ */
void setErrorMessage(const int& error) { _errorMessage = error; }
+
+ /**
+ * Read accessor to the error state
+ * @return int The error code
+ */
int getErrorMessage() { return _errorMessage; }
/**
* Get the index of the audio card for capture
- * @return _indexIn The index of the card used for capture
+ * @return int The index of the card used for capture
* 0 for the first available card on the system, 1 ...
*/
int getIndexIn() { return _indexIn; }
/**
* Get the index of the audio card for playback
- * @return _indexOut The index of the card used for playback
+ * @return int The index of the card used for playback
* 0 for the first available card on the system, 1 ...
*/
int getIndexOut() { return _indexOut; }
@@ -193,7 +232,6 @@ class AudioLayer {
* @return std::string The name of the audio plugin
*/
std::string getAudioPlugin( void ) { return _audioPlugin; }
- std::ofstream _fstream;
/**
* Get the current state. Conversation or not
* @return bool true if playSamples has been called
@@ -201,14 +239,11 @@ class AudioLayer {
*/
bool getCurrentState( void ) { return _talk; }
- int getDeviceCount();
-
/**
* Toggle echo testing on/off
*/
void toggleEchoTesting();
-
private:
/**
@@ -231,8 +266,9 @@ class AudioLayer {
/**
* Callback used for asynchronous playback.
* Called when a certain amount of data is written ot the device
+ * @param pcm_callback The callback pointer
*/
- static void AlsaCallBack( snd_async_handler_t* );
+ static void AlsaCallBack( snd_async_handler_t* pcm_callback);
/**
* Callback used for asynchronous playback.
@@ -244,8 +280,8 @@ class AudioLayer {
* Open the specified device.
* ALSA Library API
* @param pcm_p The string name for the playback device
- * pcm_c The string name for the capture device
- * flag To indicate which kind of stream you want to open
+ * @param pcm_c The string name for the capture device
+ * @param flag To indicate which kind of stream you want to open
* SFL_PCM_CAPTURE
* SFL_PCM_PLAYBACK
* SFL_PCM_BOTH
@@ -258,7 +294,7 @@ class AudioLayer {
* Copy a data buffer in the internal ring buffer
* ALSA Library API
* @param buffer The data to be copied
- * length The size of the buffer
+ * @param length The size of the buffer
* @return int The number of frames actually copied
*/
int write( void* buffer, int length);
@@ -267,7 +303,7 @@ class AudioLayer {
* Read data from the internal ring buffer
* ALSA Library API
* @param buffer The buffer to stock the read data
- * toCopy The number of bytes to get
+ * @param toCopy The number of bytes to get
* @return int The number of frames actually read
*/
int read( void* buffer, int toCopy);
@@ -277,16 +313,31 @@ class AudioLayer {
* ALSA Library API
*/
void handle_xrun_capture( void );
- void handle_xrun_playback( void );
- ManagerImpl* _manager; // augment coupling, reduce indirect access
+ /**
+ * Recover from XRUN state for playback
+ * ALSA Library API
+ */
+ void handle_xrun_playback( void );
+
+ /** Augment coupling, reduce indirect access */
+ ManagerImpl* _manager;
/**
- * Handles to manipulate capture and playback streams
+ * Handles to manipulate playback stream
* ALSA Library API
*/
snd_pcm_t* _PlaybackHandle;
+
+ /**
+ * Handles to manipulate capture stream
+ * ALSA Library API
+ */
snd_pcm_t* _CaptureHandle;
+
+ /**
+ * Alsa parameter - Size of a period in the hardware ring buffer
+ */
snd_pcm_uframes_t _periodSize;
/**
@@ -299,6 +350,12 @@ class AudioLayer {
*/
RingBuffer _urgentBuffer;
+ /**
+ * Volume is controlled by the application. Data buffer are modified here to adjust to the right volume selected by the user on the main interface
+ * @param buffer The buffer to adjust
+ * @param len The number of bytes
+ * @param stream The stream mode ( PLAYBACK - CAPTURE )
+ */
void * adjustVolume( void * , int , int);
/**
@@ -316,9 +373,13 @@ class AudioLayer {
bool deviceClosed;
/**
- * Number of audio cards on which stream has been opened
+ * Number of audio cards on which capture stream has been opened
*/
int _indexIn;
+
+ /**
+ * Number of audio cards on which playback stream has been opened
+ */
int _indexOut;
/**
@@ -341,12 +402,12 @@ class AudioLayer {
/**
* Input channel (mic) should be 1 mono
*/
- unsigned int _inChannel; // mic
+ unsigned int _inChannel;
/**
* Output channel (stereo) should be 1 mono
*/
- unsigned int _outChannel; // speaker
+ unsigned int _outChannel;
/**
* Default volume for incoming RTP and Urgent sounds.
@@ -358,12 +419,13 @@ class AudioLayer {
*/
bool _echoTesting;
+ /** Vector to manage all soundcard index - description association of the system */
std::vector<HwIDPair> IDSoundCards;
+ /** Contains the current error code */
int _errorMessage;
- ost::Mutex _mutex;
+ ost::Mutex _mutex;
};
#endif // _AUDIO_LAYER_H_
-
diff --git a/src/audio/audioloop.h b/src/audio/audioloop.h
index 647455edf7dc54a1c006e8564bda9ecaf7a4eadf..1e950c2ac75a0e0a7fe3ee0b41a428cd700accbc 100644
--- a/src/audio/audioloop.h
+++ b/src/audio/audioloop.h
@@ -26,33 +26,58 @@
#include "../global.h" // for int16 declaration
/**
- * @author Yan Morin <yan.morin@savoirfairelinux.com>
+ * @file audioloop.h
+ * @brief Loop on a sound file
*/
+
class AudioLoop {
public:
+ /**
+ * Constructor
+ */
AudioLoop();
+
+ /**
+ * Virtual destructor
+ */
virtual ~AudioLoop();
/**
- * get the next fragment of the tone
+ * Get the next fragment of the tone
* the function change the intern position, and will loop
+ * @param output The data buffer
* @param nb of int16 to send
+ * @param volume The volume
* @return the number of int16 sent (nb*2)
*/
int getNext(SFLDataFormat* output, int nb, short volume=100);
+
+ /**
+ * Reset the pointer position
+ */
void reset() { _pos = 0; }
- unsigned int getMonoSize() { return _size; }
+
+ /**
+ * Accessor to the size of the buffer
+ * @return unsigned int The size
+ */
unsigned int getSize() { return _size; }
protected:
+ /** The data buffer */
SFLDataFormat* _buffer;
- int _size; // number of int16 inside the buffer, not the delay
- int _pos; // current position, set to 0, when initialize
- int _sampleRate; // last samplerate
-};
+ /** Number of int16 inside the buffer, not the delay */
+ int _size;
+
+ /** current position, set to 0, when initialize */
+ int _pos;
+
+ /** Sample rate */
+ int _sampleRate;
+};
#endif // __AUDIOLOOP_H__
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 01f4729704b3095ac603758289e2f466b42ff9da..e98458bdc36af9fdd2098d8ffe81bb465eb94dcf 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -1,6 +1,6 @@
/*
*
- * Copyright (C) 2004-2007 Savoir-Faire Linux inc.
+ * Copyright (C) 2004-2008 Savoir-Faire Linux inc.
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
@@ -97,7 +97,6 @@ AudioRtp::closeRtpSession () {
// AudioRtpRTX Class //
////////////////////////////////////////////////////////////////////////////////
AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym)
- : _fstream("/tmp/audio.gsm", std::ofstream::binary|std::ios::out|std::ios::app)
{
setCancel(cancelDeferred);
time = new ost::Time();
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index 0947543ed07596a6be47f45a4c810d2281dcea35..0b735ac890414ed4a27b7556432d18111b0779d1 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2004-2007 Savoir-Faire Linux inc.
+ * Copyright (C) 2004-2008 Savoir-Faire Linux inc.
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
@@ -24,19 +24,20 @@
#include <cstdio>
#include <cstdlib>
-#include <fstream> // fstream + iostream for _fstream debugging...
#include <iostream>
-
#include <ccrtp/rtp.h>
#include <cc++/numbers.h>
#include <samplerate.h>
-
#include "../global.h"
#define UP_SAMPLING 0
#define DOWN_SAMPLING 1
+/**
+ * @file audiortp.h
+ * @brief Manage the real-time data transport in a SIP call
+ */
class SIPCall;
@@ -44,101 +45,170 @@ class SIPCall;
// Two pair of sockets
///////////////////////////////////////////////////////////////////////////////
class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
- public:
- AudioRtpRTX (SIPCall *, bool);
- ~AudioRtpRTX();
-
- ost::Time *time; // For incoming call notification
- virtual void run ();
-
- private:
- SIPCall* _ca;
- ost::RTPSession *_sessionSend;
- ost::RTPSession *_sessionRecv;
- ost::SymmetricRTPSession *_session;
- ost::Semaphore _start;
- bool _sym;
-
- /** When we receive data, we decode it inside this buffer */
- int16* _receiveDataDecoded;
- /** Buffers used for send data from the mic */
- unsigned char* _sendDataEncoded;
- int16* _intBufferDown;
-
- /** After that we send the data inside this buffer if there is a format conversion or rate conversion */
- /** Also use for getting mic-ringbuffer data */
- SFLDataFormat* _dataAudioLayer;
-
- /** Buffers used for sample rate conversion */
- float32* _floatBufferDown;
- float32* _floatBufferUp;
-
- /** Debugging output file */
- std::ofstream _fstream;
- FILE* fd;
-
- /** libsamplerate converter for incoming voice */
- SRC_STATE* _src_state_spkr;
- /** libsamplerate converter for outgoing voice */
- SRC_STATE* _src_state_mic;
- /** libsamplerate error */
- int _src_err;
-
- // Variables to process audio stream
- int _layerSampleRate; // sample rate for playing sound (typically 44100HZ)
- int _codecSampleRate; // sample rate of the codec we use to encode and decode (most of time 8000HZ)
- int _layerFrameSize; // length of the sound frame we capture in ms(typically 20ms)
-
- void initAudioRtpSession(void);
- /**
- * Get the data from the mic, encode it and send it through the RTP session
- */
- void sendSessionFromMic(int);
- /**
- * Get the data from the RTP packets, decode it and send it to the sound card
- */
- void receiveSessionForSpkr(int&);
- /**
- * Init the buffers used for processing sound data
- */
- void initBuffers(void);
- /**
- * Call the appropriate function, up or downsampling
- */
- int reSampleData(int, int ,int);
- /**
- * Upsample the data from the clock rate of the codec to the sample rate of the layer
- * @param int The clock rate of the codec
- * @param int The number of samples we have in the buffer
- * @return int The number of samples after the operation
- */
- int upSampleData(int, int);
- /**
- * Downsample the data from the sample rate of the layer to the clock rate of the codec
- * @param int The clock rate of the codec
- * @param int The number of samples we have in the buffer
- * @return int The number of samples after the operation
- */
- int downSampleData(int, int);
-
- AudioCodec* _audiocodec;
+ public:
+ /**
+ * Constructor
+ * @param sipcall The pointer on the SIP call
+ * @param sym Tells whether or not the voip links are symmetric
+ */
+ AudioRtpRTX (SIPCall* sipcall, bool sym);
+
+ /**
+ * Destructor
+ */
+ ~AudioRtpRTX();
+
+ /** For incoming call notification */
+ ost::Time *time;
+
+ /** Thread associated method */
+ virtual void run ();
+
+ private:
+ /** A SIP call */
+ SIPCall* _ca;
+
+ /** RTP session to send data */
+ ost::RTPSession *_sessionSend;
+
+ /** RTP session to receive data */
+ ost::RTPSession *_sessionRecv;
+
+ /** RTP symmetric session ( receive and send data in the same session ) */
+ ost::SymmetricRTPSession *_session;
+
+ /** Semaphore */
+ ost::Semaphore _start;
+
+ /** Is the session symmetric or not */
+ bool _sym;
+
+ /** When we receive data, we decode it inside this buffer */
+ int16* _receiveDataDecoded;
+
+ /** Buffers used for send data from the mic */
+ unsigned char* _sendDataEncoded;
+
+ /** Downsampled int16 buffer */
+ int16* _intBufferDown;
+
+ /** After that we send the data inside this buffer if there is a format conversion or rate conversion */
+ /** Also use for getting mic-ringbuffer data */
+ SFLDataFormat* _dataAudioLayer;
+
+ /** Downsampled float buffer */
+ float32* _floatBufferDown;
+
+ /** Upsampled float buffer */
+ float32* _floatBufferUp;
+
+ /** libsamplerate converter for incoming voice */
+ SRC_STATE* _src_state_spkr;
+
+ /** libsamplerate converter for outgoing voice */
+ SRC_STATE* _src_state_mic;
+
+ /** libsamplerate error */
+ int _src_err;
+
+ /** Variables to process audio stream: sample rate for playing sound (typically 44100HZ) */
+ int _layerSampleRate;
+
+ /** Sample rate of the codec we use to encode and decode (most of time 8000HZ) */
+ int _codecSampleRate;
+
+ /** Length of the sound frame we capture in ms(typically 20ms) */
+ int _layerFrameSize;
+
+ /**
+ * Init the RTP session. Create either symmetric or double sessions to manage data transport
+ * Set the payloads according to the manager preferences
+ */
+ void initAudioRtpSession(void);
+
+ /**
+ * Get the data from the mic, encode it and send it through the RTP session
+ * @param timestamp To manage time and synchronizing
+ */
+ void sendSessionFromMic(int timestamp);
+
+ /**
+ * Get the data from the RTP packets, decode it and send it to the sound card
+ * @param countTime To manage time and synchronizing
+ */
+ void receiveSessionForSpkr(int& countTime);
+
+ /**
+ * Init the buffers used for processing sound data
+ */
+ void initBuffers(void);
+
+ /**
+ * Call the appropriate function, up or downsampling
+ * @param sampleRate_codec The sample rate of the codec selected to encode/decode the data
+ * @param nbSamples Number of samples to process
+ * @param status Type of resampling
+ * UPSAMPLING
+ * DOWNSAMPLING
+ * @return int The number of samples after process
+ */
+ int reSampleData(int sampleRate_codec, int nbSamples, int status);
+
+ /**
+ * Upsample the data from the clock rate of the codec to the sample rate of the layer
+ * @param sampleRate_codec The sample rate of the codec selected to encode/decode the data
+ * @param nbSamples Number of samples to process
+ * @return int The number of samples after process
+ */
+ int upSampleData(int sampleRate_codec, int nbSamples);
+
+ /**
+ * Downsample the data from the sample rate of the layer to the clock rate of the codec
+ * @param sampleRate_codec The sample rate of the codec selected to encode/decode the data
+ * @param nbSamples Number of samples to process
+ * @return int The number of samples after process
+ */
+ int downSampleData(int sampleRate_codec, int nbSamples);
+
+ /** The audio codec used during the session */
+ AudioCodec* _audiocodec;
};
///////////////////////////////////////////////////////////////////////////////
// Main class rtp
///////////////////////////////////////////////////////////////////////////////
class AudioRtp {
- public:
- AudioRtp();
- ~AudioRtp();
-
- int createNewSession (SIPCall *);
- void closeRtpSession ();
-
- private:
- AudioRtpRTX* _RTXThread;
- bool _symmetric;
- ost::Mutex _threadMutex;
+ public:
+ /**
+ * Constructor
+ */
+ AudioRtp();
+
+ /**
+ * Destructor
+ */
+ ~AudioRtp();
+
+ /**
+ * Create a brand new RTP session by calling the AudioRtpRTX constructor
+ * @param ca A pointer on a SIP call
+ */
+ int createNewSession (SIPCall *ca);
+
+ /**
+ * Close a RTP session and kills the remaining threads
+ */
+ void closeRtpSession( void );
+
+ private:
+ /** The RTP thread */
+ AudioRtpRTX* _RTXThread;
+
+ /** Symmetric session or not */
+ bool _symmetric;
+
+ /** Mutex */
+ ost::Mutex _threadMutex;
};
#endif // __AUDIO_RTP_H__
diff --git a/src/audio/codecDescriptor.cpp b/src/audio/codecDescriptor.cpp
index 6ff16d5c86680cd639a2c879db8cebebaf879d32..ba6ca1e681b1219f2f61895335ac53766cb76a7a 100644
--- a/src/audio/codecDescriptor.cpp
+++ b/src/audio/codecDescriptor.cpp
@@ -26,10 +26,6 @@
CodecDescriptor::CodecDescriptor()
{
- //init();
- //#ifdef HAVE_SPEEX
- //_codecMap[PAYLOAD_CODEC_SPEEX] = new CodecSpeex(PAYLOAD_CODEC_SPEEX); // TODO: this is a variable payload!
- //#endif
}
CodecDescriptor::~CodecDescriptor()
@@ -54,7 +50,6 @@ CodecDescriptor::init()
_nbCodecs = CodecDynamicList.size();
if( _nbCodecs <= 0 ){
_debug(" Error - No codecs available in directory %s\n" , CODECS_DIR);
- //exit(0);
}
int i;
diff --git a/src/audio/codecDescriptor.h b/src/audio/codecDescriptor.h
index fd5775fa8d52fb7392b60ca5753b4bfa2a9d009a..92093cd32ee33034df0870e43476233f8d463ea6 100644
--- a/src/audio/codecDescriptor.h
+++ b/src/audio/codecDescriptor.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+ * Copyright (C) 2004-2008 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
@@ -19,37 +19,36 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-
#ifndef __CODEC_DESCRIPTOR_H__
#define __CODEC_DESCRIPTOR_H__
#include <string>
#include <map>
#include <vector>
-// To read directories content
#include <dirent.h>
#include "../global.h"
#include "../user_cfg.h"
#include "codecs/audiocodec.h"
+/** Enumeration that contains known audio payloads */
typedef enum {
-// http://www.iana.org/assignments/rtp-parameters
-// http://www.gnu.org/software/ccrtp/doc/refman/html/formats_8h.html#a0
-// 0 PCMU A 8000 1 [RFC3551]
+ // http://www.iana.org/assignments/rtp-parameters
+ // http://www.gnu.org/software/ccrtp/doc/refman/html/formats_8h.html#a0
+ // 0 PCMU A 8000 1 [RFC3551]
PAYLOAD_CODEC_ULAW = 0,
-// 3 GSM A 8000 1 [RFC3551]
+ // 3 GSM A 8000 1 [RFC3551]
PAYLOAD_CODEC_GSM = 3,
-// 8 PCMA A 8000 1 [RFC3551]
+ // 8 PCMA A 8000 1 [RFC3551]
PAYLOAD_CODEC_ALAW = 8,
-// http://www.ietf.org/rfc/rfc3952.txt
-// 97 iLBC/8000
+ // http://www.ietf.org/rfc/rfc3952.txt
+ // 97 iLBC/8000
PAYLOAD_CODEC_ILBC_20 = 97,
PAYLOAD_CODEC_ILBC_30 = 98,
-// http://www.speex.org/drafts/draft-herlein-speex-rtp-profile-00.txt
-// 97 speex/8000
-// http://support.xten.com/viewtopic.php?p=8684&sid=3367a83d01fdcad16c7459a79859b08e
-// 100 speex/16000
+ // http://www.speex.org/drafts/draft-herlein-speex-rtp-profile-00.txt
+ // 97 speex/8000
+ // http://support.xten.com/viewtopic.php?p=8684&sid=3367a83d01fdcad16c7459a79859b08e
+ // 100 speex/16000
PAYLOAD_CODEC_SPEEX_8000 = 110,
PAYLOAD_CODEC_SPEEX_16000 = 111,
PAYLOAD_CODEC_SPEEX_32000 = 112
@@ -61,200 +60,212 @@ typedef enum {
typedef std::map<AudioCodecType, std::string> CodecMap;
/** The struct to reflect the order the user wants to use the codecs */
typedef std::vector<AudioCodecType> CodecOrder;
-
+/** Enable us to keep the handle pointer on the codec dynamicaly loaded so that we could destroy when we dont need it anymore */
typedef std::pair<AudioCodec* , void*> CodecHandlePointer;
+/** Maps a pointer on an audiocodec object to a payload */
typedef std::map<AudioCodecType , AudioCodec*> CodecsMap;
-class CodecDescriptor {
-public:
- /**
- * Initialize all codec
- */
- CodecDescriptor();
- ~CodecDescriptor();
-
- /**
- * Accessor to data structures
- */
- CodecsMap& getCodecsMap() { return _CodecsMap; }
- CodecOrder& getActiveCodecs() { return _codecOrder; }
-
- /**
- * Get codec name by its payload
- * @param payload the payload looked for
- * same as getPayload()
- * @return the name of the codec
- */
- std::string getCodecName(AudioCodecType payload);
-
- /**
- * Get the codec object associated with the payload
- * @param payload The payload looked for
- * @return AudioCodec* A pointer on a AudioCodec object
- */
- AudioCodec* getCodec( AudioCodecType payload );
-
- /**
- * Initialiaze the map with all the supported codecs, even those inactive
- */
- void init();
-
- /**
- * Set the default codecs order
- */
- void setDefaultOrder();
-
- /**
- * Check in the map codec if the specified codec is supported
- * @param payload unique identifier of a codec (RFC)
- * @return true if the codec specified is supported
- * false otherwise
- */
- bool isActive(AudioCodecType payload);
-
- /**
- * Remove the codec with payload payload from the list
- * @param payload the codec to erase
- */
- void removeCodec(AudioCodecType payload);
-
- /**
- * Add a codec in the list.
- * @param payload the codec to add
- */
- void addCodec(AudioCodecType payload);
-
- /**
- * Get the bit rate of the specified codec.
- * @param payload The payload of the codec
- * @return double The bit rate
- */
- double getBitRate(AudioCodecType payload);
-
- /**
- * Get the bandwidth for one call with the specified codec.
- * The value has been calculated with the further information:
- * RTp communication, SIP protocol (the value with IAX2 is very close), no RTCP, one simultaneous call, for one channel (the incoming one).
- * @param payload The payload of the codec
- * @return double The bandwidth
- */
- double getBandwidthPerCall(AudioCodecType payload);
-
-
- /**
- * Get the clock rate of the specified codec
- * @param payload The payload of the codec
- * @return int The clock rate of the specified codec
- */
- int getSampleRate(AudioCodecType payload);
-
- /**
- * Get the number of channels
- * @param payload The payload of the codec
- * @return int Number of channels
- */
- int getChannel(AudioCodecType payload);
-
-/**
- * Set the order of codecs by their payload
- * @param list The ordered list sent by DBus
+/*
+ * @file codecdescriptor.h
+ * @brief Handle audio codecs, load them in memory
*/
- void saveActiveCodecs(const std::vector<std::string>& list);
-
-
- std::string getDescription( std::string );
- /**
- * Get the number of codecs loaded in dynamic memory
- */
- int getCodecsNumber( void ) { return _nbCodecs; }
-
- /**
- * Unreferences the codecs loaded in memory
- */
- void deleteHandlePointer( void );
-
- /**
- * Get the first element of the CodecsMap struct.
- * i.e the one with the lowest payload
- * @return AudioCodec The pointer on the codec object
- */
- AudioCodec* getFirstCodecAvailable( void );
-
-private:
-
- /**
- * Scan the installation directory ( --prefix configure option )
- * And load the dynamic library
- * @return std::vector<AudioCodec*> The list of the codec object successfully loaded in memory
- */
- std::vector<AudioCodec *> scanCodecDirectory( void );
-
- /**
- * Load a codec
- * @param std::string The path of the shared ( dynamic ) library.
- * @return AudioCodec* the pointer of the object loaded.
- */
- AudioCodec* loadCodec( std::string );
-
- /**
- * Unload a codec
- * @param CodecHandlePointer The map containing the pointer on the object and the pointer on the handle function
- */
- void unloadCodec( CodecHandlePointer );
-
- /**
- * Check if the files found in searched directories seems valid
- * @param std::string The name of the file
- * @return true if the file name begins with libcodec_ and ends with .so
- * false otherwise
- */
- bool seemsValid( std::string );
-
- /**
- * Check if the codecs shared library has already been scanned during the session
- * Useful not to load twice the same codec saved in the different directory
- * @param std::string The complete name of the shared directory ( without the path )
- * @return true if the codecs has been scanned
- * false otherwise
- */
- bool alreadyInCache( std::string );
-
- /**
- * Check if the audiocodec object has been successfully created
- * @param payload The payload of the codec
- * @return true if the audiocodec has been created
- * false otherwise
- */
- bool isCodecLoaded( int payload );
-
- /**
- * Map the payload of a codec and the object associated ( AudioCodec * )
- */
- CodecsMap _CodecsMap;
-
- /**
- * Vector containing the order of the codecs
- */
- CodecOrder _codecOrder;
-
- /**
- * Vector containing the complete name of the codec shared library scanned
- */
- std::vector<std::string> _Cache;
-
- /**
- * Number of codecs loaded
- */
- int _nbCodecs;
-
- /**
- * Vector containing pairs
- * Pair between pointer on function handle and pointer on audiocodec object
- */
- std::vector< CodecHandlePointer > _CodecInMemory;
+class CodecDescriptor {
+ public:
+ /**
+ * Constructor
+ */
+ CodecDescriptor();
+
+ /**
+ * Destructor
+ */
+ ~CodecDescriptor();
+
+ /**
+ * Accessor to data structures
+ * @return CodecsMap& The available codec
+ */
+ CodecsMap& getCodecsMap() { return _CodecsMap; }
+
+ /**
+ * Accessor to data structures
+ * @return CodecOrder& The list that reflects the user's choice
+ */
+ CodecOrder& getActiveCodecs() { return _codecOrder; }
+
+ /**
+ * Get codec name by its payload
+ * @param payload the payload looked for
+ * same as getPayload()
+ * @return std::string The name of the codec
+ */
+ std::string getCodecName(AudioCodecType payload);
+
+ /**
+ * Get the codec object associated with the payload
+ * @param payload The payload looked for
+ * @return AudioCodec* A pointer on a AudioCodec object
+ */
+ AudioCodec* getCodec( AudioCodecType payload );
+
+ /**
+ * Initialiaze the map with all the supported codecs, even those inactive
+ */
+ void init();
+
+ /**
+ * Set the default codecs order
+ */
+ void setDefaultOrder();
+
+ /**
+ * Check in the map codec if the specified codec is supported
+ * @param payload unique identifier of a codec (RFC)
+ * @return bool True if the codec specified is supported
+ * false otherwise
+ */
+ bool isActive(AudioCodecType payload);
+
+ /**
+ * Remove the codec with payload payload from the list
+ * @param payload the codec to erase
+ */
+ void removeCodec(AudioCodecType payload);
+
+ /**
+ * Add a codec in the list.
+ * @param payload the codec to add
+ */
+ void addCodec(AudioCodecType payload);
+
+ /**
+ * Get the bit rate of the specified codec.
+ * @param payload The payload of the codec
+ * @return double The bit rate
+ */
+ double getBitRate(AudioCodecType payload);
+
+ /**
+ * Get the bandwidth for one call with the specified codec.
+ * The value has been calculated with the further information:
+ * RTp communication, SIP protocol (the value with IAX2 is very close), no RTCP, one simultaneous call, for one channel (the incoming one).
+ * @param payload The payload of the codec
+ * @return double The bandwidth
+ */
+ double getBandwidthPerCall(AudioCodecType payload);
-};
+ /**
+ * Get the clock rate of the specified codec
+ * @param payload The payload of the codec
+ * @return int The clock rate of the specified codec
+ */
+ int getSampleRate(AudioCodecType payload);
+
+ /**
+ * Get the number of channels
+ * @param payload The payload of the codec
+ * @return int Number of channels
+ */
+ int getChannel(AudioCodecType payload);
+
+ /**
+ * Set the order of codecs by their payload
+ * @param list The ordered list sent by DBus
+ */
+ void saveActiveCodecs(const std::vector<std::string>& list);
+
+ /**
+ * Get the number of codecs loaded in dynamic memory
+ * @return int The number
+ */
+ int getCodecsNumber( void ) { return _nbCodecs; }
+
+ /**
+ * Unreferences the codecs loaded in memory
+ */
+ void deleteHandlePointer( void );
+
+ /**
+ * Get the first element of the CodecsMap struct.
+ * i.e the one with the lowest payload
+ * @return AudioCodec The pointer on the codec object
+ */
+ AudioCodec* getFirstCodecAvailable( void );
+
+ private:
+
+ /**
+ * Scan the installation directory ( --prefix configure option )
+ * And load the dynamic library
+ * @return std::vector<AudioCodec*> The list of the codec object successfully loaded in memory
+ */
+ std::vector<AudioCodec *> scanCodecDirectory( void );
+
+ /**
+ * Load a codec
+ * @param std::string The path of the shared ( dynamic ) library.
+ * @return AudioCodec* the pointer of the object loaded.
+ */
+ AudioCodec* loadCodec( std::string );
+
+ /**
+ * Unload a codec
+ * @param CodecHandlePointer The map containing the pointer on the object and the pointer on the handle function
+ */
+ void unloadCodec( CodecHandlePointer );
+
+ /**
+ * Check if the files found in searched directories seems valid
+ * @param std::string The name of the file
+ * @return bool True if the file name begins with libcodec_ and ends with .so
+ * false otherwise
+ */
+ bool seemsValid( std::string );
+
+ /**
+ * Check if the codecs shared library has already been scanned during the session
+ * Useful not to load twice the same codec saved in the different directory
+ * @param std::string The complete name of the shared directory ( without the path )
+ * @return bool True if the codecs has been scanned
+ * false otherwise
+ */
+ bool alreadyInCache( std::string );
+
+ /**
+ * Check if the audiocodec object has been successfully created
+ * @param payload The payload of the codec
+ * @return bool True if the audiocodec has been created
+ * false otherwise
+ */
+ bool isCodecLoaded( int payload );
+
+ /**
+ * Map the payload of a codec and the object associated ( AudioCodec * )
+ */
+ CodecsMap _CodecsMap;
+
+ /**
+ * Vector containing the order of the codecs
+ */
+ CodecOrder _codecOrder;
+
+ /**
+ * Vector containing the complete name of the codec shared library scanned
+ */
+ std::vector<std::string> _Cache;
+
+ /**
+ * Number of codecs loaded
+ */
+ int _nbCodecs;
+
+ /**
+ * Vector containing pairs
+ * Pair between pointer on function handle and pointer on audiocodec object
+ */
+ std::vector< CodecHandlePointer > _CodecInMemory;
+};
#endif // __CODEC_DESCRIPTOR_H__
diff --git a/src/audio/common.h b/src/audio/common.h
index fb2cb85c834069230c1f350bb2ed82047d00146e..5be3d28e17ef2af2cf2548cc4d4d037e0436f953 100644
--- a/src/audio/common.h
+++ b/src/audio/common.h
@@ -41,12 +41,10 @@ typedef signed short int16; /**< An 16 bit signed integer (2s complement) */
typedef signed int int32; /**< An 32 bit signed integer (2s complement) */
typedef unsigned int uint; /**< An unsigned integer or at least 32 bits */
-
#ifndef NULL
#define NULL 0 /**< Used to represent a null pointer type */
#endif
-
#ifdef _MSC_VER // Compiling for Microsoft Visual C++
#define DEBUGGER { _asm int 3 } /**< Invoke debugger */
@@ -59,17 +57,13 @@ typedef unsigned int uint; /**< An unsigned integer or at least 32 bits */
#define DEBUGGER /**< Invoke debugger */
#define IMPORT /**< Mark a function which is to be imported from a DLL */
#define EXPORT /**< Mark a function to be exported from a DLL */
-//#define ASSERT(c) /**< Assert that expression 'c' is true */
#endif
-
#ifdef DEBUG
#define ASSERT_DEBUG(c) ASSERT(c) /**< Assert that expression 'c' is true (when compiled for debugging)*/
#else
#define ASSERT_DEBUG(c)
#endif
-
#endif
-
diff --git a/src/audio/dtmf.h b/src/audio/dtmf.h
index b3713eadd050fbd404c279779ed55dc377a1a61c..9041514d48feaa2c70ce9a043f2b21b77b823108 100644
--- a/src/audio/dtmf.h
+++ b/src/audio/dtmf.h
@@ -28,29 +28,39 @@
#include "dtmfgenerator.h"
/**
- * DMTF library to generate a dtmf sample
+ * @file dtmf.h
+ * @brief DMTF library to generate a dtmf sample
*/
class DTMF {
-public:
- /**
- * Create a new DTMF.
- * @param samplingRate frequency of the sample (ex: 8000 hz)
- */
- DTMF (unsigned int sampleRate);
- ~DTMF (void);
-
- void startTone (char);
- /**
- * Copy the sound inside the sampling* buffer
- * @param buffer : a SFLDataFormat* buffer
- * @param n :
- */
- bool generateDTMF (SFLDataFormat* buffer, size_t n);
+ public:
+ /**
+ * Create a new DTMF.
+ * @param samplingRate frequency of the sample (ex: 8000 hz)
+ */
+ DTMF (unsigned int sampleRate);
+
+ /**
+ * Destructor
+ */
+ ~DTMF (void);
- char currentTone;
- char newTone;
+ /**
+ * Start the done for th given dtmf
+ * @param code The DTMF code
+ */
+ void startTone(char code);
+
+ /**
+ * Copy the sound inside the sampling* buffer
+ * @param buffer : a SFLDataFormat* buffer
+ * @param n : The size to generate
+ */
+ bool generateDTMF (SFLDataFormat* buffer, size_t n);
- DTMFGenerator dtmfgenerator;
+ char currentTone;
+ char newTone;
+
+ DTMFGenerator dtmfgenerator;
};
#endif // __KEY_DTMF_H_
diff --git a/src/audio/dtmfgenerator.h b/src/audio/dtmfgenerator.h
index 71b25852e513112f270142914b908f185e8df009..376d31ce82b6b93ec974127b4f1c2edb87f7c4e2 100644
--- a/src/audio/dtmfgenerator.h
+++ b/src/audio/dtmfgenerator.h
@@ -28,85 +28,108 @@
#include <exception>
#include <string.h>
-//#include "tonegenerator.h"
#include "tone.h"
#define NUM_TONES 16
/*
- * DMTF Generator Exception
+ * @file dtmfgenerator.h
+ * @brief DMTF Generator Exception
*/
class DTMFException : public std::exception
{
-private:
- const char* reason;
-public:
- DTMFException(const char* _reason) throw();
- virtual ~DTMFException() throw();
- virtual const char* what() const throw();
+ private:
+ /** Message */
+ const char* reason;
+ public:
+ /**
+ * Constructor
+ * @param _reason An error message
+ */
+ DTMFException(const char* _reason) throw();
+
+ /**
+ * Destructor
+ */
+ virtual ~DTMFException() throw();
+
+ /**
+ * @return const char* The error
+ */
+ virtual const char* what() const throw();
};
-
/*
- * DTMF Tone Generator
+ * @file dtmfgenerator.h
+ * @brief DTMF Tone Generator
*/
class DTMFGenerator
{
-private:
- struct DTMFTone {
- unsigned char code; // Code of the tone
- int lower; // Lower frequency
- int higher; // Higher frequency
- };
-
-/*
- * State of the DTMF generator
- */
- struct DTMFState {
- unsigned int offset; // Offset in the sample currently being played
- SFLDataFormat* sample; // Currently generated code
- };
-
- DTMFState state;
- static const DTMFTone tones[NUM_TONES];
-
- SFLDataFormat* samples[NUM_TONES]; // Generated samples
-
- /**
- * Sampling rate of generated dtmf
- */
- int _sampleRate;
- Tone tone;
-
-public:
- /**
- * DTMF Generator contains frequency of each keys
- * and can build one DTMF.
- * @param sampleRate frequency of the sample (ex: 8000 hz)
- */
- DTMFGenerator(unsigned int sampleRate);
- ~DTMFGenerator();
-
-/*
- * Get n samples of the signal of code code
- * @param buffer a SFLDataFormat pointer to an allocated buffer
- * @param n number of sampling to get, should be lower or equal to buffer size
- * @parma code dtmf code to get sound
- */
- void getSamples(SFLDataFormat* buffer, size_t n, unsigned char code) throw (DTMFException);
-
-/*
- * Get next n samples (continues where previous call to
- * genSample or genNextSamples stopped
- * @param buffer a SFLDataFormat pointer to an allocated buffer
- * @param n number of sampling to get, should be lower or equal to buffer size
- */
- void getNextSamples(SFLDataFormat* buffer, size_t n) throw (DTMFException);
-
-private:
- SFLDataFormat* generateSample(unsigned char code) throw (DTMFException);
-
+ private:
+ /** Struct to handle a DTMF */
+ struct DTMFTone {
+ unsigned char code; /** Code of the tone */
+ int lower; /** Lower frequency */
+ int higher; /** Higher frequency */
+ };
+
+ /** State of the DTMF generator */
+ struct DTMFState {
+ unsigned int offset; /** Offset in the sample currently being played */
+ SFLDataFormat* sample; /** Currently generated code */
+ };
+
+ /** State of the DTMF generator */
+ DTMFState state;
+
+ /** The different kind of tones */
+ static const DTMFTone tones[NUM_TONES];
+
+ /** Generated samples */
+ SFLDataFormat* samples[NUM_TONES];
+
+ /** Sampling rate of generated dtmf */
+ int _sampleRate;
+
+ /** A tone object */
+ Tone tone;
+
+ public:
+ /**
+ * DTMF Generator contains frequency of each keys
+ * and can build one DTMF.
+ * @param sampleRate frequency of the sample (ex: 8000 hz)
+ */
+ DTMFGenerator(unsigned int sampleRate);
+
+ /**
+ * Destructor
+ */
+ ~DTMFGenerator();
+
+ /*
+ * Get n samples of the signal of code code
+ * @param buffer a SFLDataFormat pointer to an allocated buffer
+ * @param n number of sampling to get, should be lower or equal to buffer size
+ * @param code dtmf code to get sound
+ */
+ void getSamples(SFLDataFormat* buffer, size_t n, unsigned char code) throw (DTMFException);
+
+ /*
+ * Get next n samples (continues where previous call to
+ * genSample or genNextSamples stopped
+ * @param buffer a SFLDataFormat pointer to an allocated buffer
+ * @param n number of sampling to get, should be lower or equal to buffer size
+ */
+ void getNextSamples(SFLDataFormat* buffer, size_t n) throw (DTMFException);
+
+ private:
+ /**
+ * Generate samples for a specific dtmf code
+ * @param code The code
+ * @return SFLDataFormat* The generated data
+ */
+ SFLDataFormat* generateSample(unsigned char code) throw (DTMFException);
};
-
#endif // DTMFGENERATOR_H
diff --git a/src/audio/ringbuffer.h b/src/audio/ringbuffer.h
index 1a06c8e723b849af13784ed60e59a8efd60efc5e..b6ae54473a8e3db2d8f44f83bb1fba48c4fba3a0 100644
--- a/src/audio/ringbuffer.h
+++ b/src/audio/ringbuffer.h
@@ -26,40 +26,81 @@
typedef unsigned char* samplePtr;
class RingBuffer {
- public:
- RingBuffer(int size);
- ~RingBuffer();
+ public:
+ /**
+ * Constructor
+ * @param size Size of the buffer to create
+ */
+ RingBuffer(int size);
- // To set counters to 0
- void flush (void);
-
- //
- // For the writer only:
- //
- int AvailForPut (void) const;
- int Put (void*, int, unsigned short volume = 100);
+ /**
+ * Destructor
+ */
+ ~RingBuffer();
- //
- // For the reader only:
- //
- int AvailForGet (void) const;
- int Get (void *, int, unsigned short volume = 100);
- int Discard(int);
+ /**
+ * Reset the counters to 0
+ */
+ void flush (void);
- int Len() const;
- /**
- * Debug function print mEnd, mStart, mBufferSize
- */
- void debug();
-
- private:
+ /**
+ * To get how much space is available in the buffer to write in
+ * @return int The available size
+ */
+ int AvailForPut (void) const;
- int mStart;
- int mEnd;
- int mBufferSize;
- samplePtr mBuffer;
-};
+ /**
+ * Write data in the ring buffer
+ * @param buffer Data to copied
+ * @param toCopy Number of bytes to copy
+ * @param volume The volume
+ * @return int Number of bytes copied
+ */
+ int Put (void* buffer, int toCopy, unsigned short volume = 100);
-#endif /* __RING_BUFFER__ */
+ /**
+ * To get how much space is available in the buffer to read in
+ * @return int The available size
+ */
+ int AvailForGet (void) const;
+
+ /**
+ * Get data in the ring buffer
+ * @param buffer Data to copied
+ * @param toCopy Number of bytes to copy
+ * @param volume The volume
+ * @return int Number of bytes copied
+ */
+ int Get (void* buffer, int toCopy, unsigned short volume = 100);
+
+ /**
+ * Discard data from the buffer
+ * @param toDiscard Number of bytes to discard
+ * @return int Number of bytes discarded
+ */
+ int Discard(int toDiscard);
+ /**
+ * Total length of the ring buffer
+ * @return int
+ */
+ int Len() const;
+
+ /**
+ * Debug function print mEnd, mStart, mBufferSize
+ */
+ void debug();
+ private:
+
+ /** Pointer on the first data */
+ int mStart;
+ /** Pointer on the last data */
+ int mEnd;
+ /** Buffer size */
+ int mBufferSize;
+ /** Data */
+ samplePtr mBuffer;
+};
+
+#endif /* __RING_BUFFER__ */
diff --git a/src/audio/tone.h b/src/audio/tone.h
index a9de7a3d34fd351edab86ce735cd3c34e97582c3..e75a865e7d2fd45287514e892c144f3565fe0181 100644
--- a/src/audio/tone.h
+++ b/src/audio/tone.h
@@ -29,17 +29,24 @@
#define TONE_NBCOUNTRY 7
/**
- * Tone sample (dial, busy, ring, congestion)
- * @author Yan Morin <yan.morin@savoirfairelinux.com>
+ * @file tone.h
+ * @brief Tone sample (dial, busy, ring, congestion)
*/
class Tone : public AudioLoop {
public:
/**
+ * Constructor
* @param definition String that contain frequency/time of the tone
* @param sampleRate SampleRating of audio tone
*/
Tone(const std::string& definition, unsigned int sampleRate);
+
+ /**
+ * Destructor
+ */
~Tone();
+
+ /** The different kind of tones */
enum TONEID {
TONE_DIALTONE = 0,
TONE_BUSY,
@@ -47,8 +54,12 @@ public:
TONE_CONGESTION,
TONE_NULL
};
+
/**
- * add a simple or double sin to the buffer, it double the sin in stereo
+ * Add a simple or double sin to the buffer, it double the sin in stereo
+ * @param buffer The data
+ * @param frequency1 The first frequency
+ * @param frequency2 The second frequency
* @param nb are the number of int16 (mono) to generate
* by example nb=5 generate 10 int16, 5 for the left, 5 for the right
*/
@@ -58,9 +69,11 @@ private:
/**
* allocate the memory with the definition
+ * @param definition String that contain frequency/time of the tone.
*/
void genBuffer(const std::string& definition);
+ /** Sample rate */
unsigned int _sampleRate;
};
diff --git a/src/audio/tonegenerator.cpp b/src/audio/tonegenerator.cpp
index 2ef0ac0b276bcce60700e3c3a3fdda5d6f2ceb53..37cfd1eaaab1def74be072e130876ce41b129873 100644
--- a/src/audio/tonegenerator.cpp
+++ b/src/audio/tonegenerator.cpp
@@ -41,9 +41,6 @@ ToneGenerator::~ToneGenerator (void) {
/**
* Calculate superposition of 2 sinus
*
- * @param lower frequency
- * @param higher frequency
- * @param ptr for result buffer
*/
void
ToneGenerator::generateSin (int lowerfreq, int higherfreq, int16* ptr, int len) const {
diff --git a/src/audio/tonegenerator.h b/src/audio/tonegenerator.h
index b649398378e4d31b6d33b51e8872ead688e45c7b..062caebd518387c129a363a70702ce454331e764 100644
--- a/src/audio/tonegenerator.h
+++ b/src/audio/tonegenerator.h
@@ -27,35 +27,49 @@
#include "../global.h"
/**
- * Sine generator to create tone with string definition
+ * @file tonegenerator.h
+ * @brief Sine generator to create tone with string definition
*/
+
class ToneGenerator {
-public:
- ToneGenerator (unsigned int sampleRate);
- ~ToneGenerator (void);
-
- /**
- * Calculate sinus with superposition of 2 frequencies
- */
- void generateSin (int, int, int16 *, int len) const;
-
-
- ///////////////////////////
- // Public members variable
- //////////////////////////
- int16 *sample;
- int freq1, freq2;
- int time;
- int totalbytes;
-
-private:
- /*
- * Initialisation of the supported tones according to the countries.
- */
- void initTone (void);
-
- int16 _buf[SIZEBUF];
- int _sampleRate;
+ public:
+ /**
+ * Constructor
+ * @param sampleRate The sample rate of the generated samples
+ */
+ ToneGenerator (unsigned int sampleRate);
+
+ /**
+ * Destructor
+ */
+ ~ToneGenerator (void);
+
+ /**
+ * Calculate sinus with superposition of 2 frequencies
+ * @param lowerfreq Lower frequency
+ * @param higherfreq Higher frequency
+ * @param ptr For result buffer
+ * @param len The length of the data to be generated
+ */
+ void generateSin (int, int, int16 *, int len) const;
+
+
+ ///////////////////////////
+ // Public members variable
+ //////////////////////////
+ int16 *sample;
+ int freq1, freq2;
+ int time;
+ int totalbytes;
+
+ private:
+ /*
+ * Initialisation of the supported tones according to the countries.
+ */
+ void initTone (void);
+
+ int16 _buf[SIZEBUF];
+ int _sampleRate;
};
#endif // __TONE_GENRATOR_H__
diff --git a/src/audio/tonelist.h b/src/audio/tonelist.h
index 20d681ccac17a4f4b59f8dc2bd6d3103a2151ce1..a381193c025b9d4cad54a4a527bbdc84c34397ed 100644
--- a/src/audio/tonelist.h
+++ b/src/audio/tonelist.h
@@ -25,13 +25,22 @@
#include "tone.h"
/**
- * @author Yan Morin <yan.morin@savoirfairelinux.com>
+ * @file tonelist.h
+ * @brief Manages the different kind of tones according to the country
*/
class ToneList {
public:
+ /**
+ * Constructor
+ */
ToneList();
+
+ /**
+ * Destructor
+ */
~ToneList();
+ /** Countries */
enum COUNTRYID {
ZID_NORTH_AMERICA = 0,
ZID_FRANCE,
@@ -43,22 +52,26 @@ public:
};
/**
- * get the string definition of a tone
+ * Get the string definition of a tone
* return the default country or default tone if id are invalid
- * @param country the country Id, see ToneList constructor for the list
- * @param toneId toneId
- * @return a string definition of the tone
+ * @param countryId The country Id, see ToneList constructor for the list
+ * @param toneId The toneId
+ * @return std::string A string definition of the tone
*/
std::string getDefinition(COUNTRYID countryId, Tone::TONEID toneId);
+
/**
- * get the country id associate to a country name
+ * Get the country id associate to a country name
* return the default country id if not found
* The default tone/country are set inside the ToneList constructor
* @param countryName countryName, see the ToneList constructor list
- * @return Country Id or default Id
+ * @return COUNTRYID Country Id or default Id
*/
COUNTRYID getCountryId(const std::string& countryName);
+
+ /** @return int The number of tones */
int getNbTone() { return _nbTone; }
+
private:
void initToneDefinition();
std::string _toneZone[TONE_NBCOUNTRY][TONE_NBTONE];
diff --git a/src/iaxaccount.h b/src/iaxaccount.h
index 2cdfb6232ad0dc338cdb708d69bfdbb7ed34b7f2..0d52718e5791b8c76fe4306614a579a78b5ca8ce 100644
--- a/src/iaxaccount.h
+++ b/src/iaxaccount.h
@@ -25,7 +25,7 @@
/**
* @file: iaxaccount.h
- * @brief: an IAX Account specify IAX specific functions and objects (IAXCall/IAXVoIPLink)
+ * @brief An IAX Account specify IAX specific functions and objects (IAXCall/IAXVoIPLink)
*/
class IAXAccount : public Account
{
diff --git a/src/iaxcall.h b/src/iaxcall.h
index ccfe8c41d5dd3e69b9ba9ef1089cf52c0b225a8f..1fdf5d75e68c78ba1b0721fa39bd532bff355e28 100644
--- a/src/iaxcall.h
+++ b/src/iaxcall.h
@@ -26,7 +26,7 @@
/**
* @file: iaxcall.h
- * @brief: IAXCall are IAX implementation of a normal Call
+ * @brief IAXCall are IAX implementation of a normal Call
*/
class IAXCall : public Call