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update v2.x

This commit is contained in:
bossiel 2011-05-03 18:26:29 +00:00
parent f60d8c512c
commit bbb67ae87e
38 changed files with 2599 additions and 159 deletions
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130
branches/2.0/doubango/bindings/_common/SipStack.cxx
View File

@ -44,11 +44,11 @@ static int session_handle_event(const tsip_event_t *sipevent);
//# endif
#endif
SipStack::SipStack(SipCallback* callback_, const char* realm_uri, const char* impi_uri, const char* impu_uri)
SipStack::SipStack(SipCallback* pCallback, const char* realm_uri, const char* impi_uri, const char* impu_uri)
:SafeObject()
{
this->debugCallback = tsk_null;
this->callback = callback_;
m_pDebugCallback = tsk_null;
m_pCallback = pCallback;
/* Initialize network layer */
if(SipStack::count == 0){
@ -57,7 +57,7 @@ SipStack::SipStack(SipCallback* callback_, const char* realm_uri, const char* im
}
/* Creates stack handle */
this->handle = tsip_stack_create(stack_callback, realm_uri, impi_uri, impu_uri,
m_pHandle = tsip_stack_create(stack_callback, realm_uri, impi_uri, impu_uri,
TSIP_STACK_SET_LOCAL_IP(DEFAULT_LOCAL_IP),
TSIP_STACK_SET_USERDATA(this), /* used as context (useful for server-initiated requests) */
TSIP_STACK_SET_NULL());
@ -70,7 +70,7 @@ SipStack::~SipStack()
this->stop();
/* Destroy stack handle */
TSK_OBJECT_SAFE_FREE(this->handle);
TSK_OBJECT_SAFE_FREE(m_pHandle);
/* DeInitialize the network layer (only if last stack) */
if(--SipStack::count == 0){
@ -81,13 +81,13 @@ SipStack::~SipStack()
bool SipStack::start()
{
return (tsip_stack_start(this->handle) == 0);
return (tsip_stack_start(m_pHandle) == 0);
}
bool SipStack::setDebugCallback(DDebugCallback* callback)
bool SipStack::setDebugCallback(DDebugCallback* pCallback)
{
if(this && callback){
this->debugCallback = callback;
if(this && pCallback){
m_pDebugCallback = pCallback;
tsk_debug_set_arg_data(this);
tsk_debug_set_info_cb(DDebugCallback::debug_info_cb);
tsk_debug_set_warn_cb(DDebugCallback::debug_warn_cb);
@ -95,7 +95,7 @@ bool SipStack::setDebugCallback(DDebugCallback* callback)
tsk_debug_set_fatal_cb(DDebugCallback::debug_fatal_cb);
}
else if(this){
this->debugCallback = tsk_null;
m_pDebugCallback = tsk_null;
tsk_debug_set_arg_data(tsk_null);
tsk_debug_set_info_cb(tsk_null);
tsk_debug_set_warn_cb(tsk_null);
@ -108,7 +108,7 @@ bool SipStack::setDebugCallback(DDebugCallback* callback)
bool SipStack::setRealm(const char* realm_uri)
{
int ret = tsip_stack_set(this->handle,
int ret = tsip_stack_set(m_pHandle,
TSIP_STACK_SET_REALM(realm_uri),
TSIP_STACK_SET_NULL());
return (ret == 0);
@ -116,21 +116,21 @@ bool SipStack::setRealm(const char* realm_uri)
bool SipStack::setIMPI(const char* impi)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_IMPI(impi),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setIMPU(const char* impu_uri)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_IMPU(impu_uri),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setPassword(const char* password)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_PASSWORD(password),
TSIP_STACK_SET_NULL()) == 0);
}
@ -138,14 +138,14 @@ bool SipStack::setPassword(const char* password)
bool SipStack::setAMF(const char* amf)
{
uint16_t _amf = (uint16_t)tsk_atox(amf);
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_IMS_AKA_AMF(_amf),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setOperatorId(const char* opid)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_IMS_AKA_OPERATOR_ID(opid),
TSIP_STACK_SET_NULL()) == 0);
}
@ -153,14 +153,14 @@ bool SipStack::setOperatorId(const char* opid)
bool SipStack::setProxyCSCF(const char* fqdn, unsigned short port, const char* transport, const char* ipversion)
{
unsigned _port = port;//promote
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_PROXY_CSCF(fqdn, _port, transport, ipversion),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setLocalIP(const char* ip)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_LOCAL_IP(ip),
TSIP_STACK_SET_NULL()) == 0);
}
@ -168,34 +168,34 @@ bool SipStack::setLocalIP(const char* ip)
bool SipStack::setLocalPort(unsigned short port)
{
unsigned _port = port;//promote
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_LOCAL_PORT(_port),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setEarlyIMS(bool enabled){
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_EARLY_IMS(enabled? tsk_true : tsk_false),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::addHeader(const char* name, const char* value)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_HEADER(name, value),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::removeHeader(const char* name)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_UNSET_HEADER(name),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::addDnsServer(const char* ip)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_DNS_SERVER(ip),
TSIP_STACK_SET_NULL()) == 0);
}
@ -203,21 +203,21 @@ bool SipStack::addDnsServer(const char* ip)
bool SipStack::setDnsDiscovery(bool enabled)
{
tsk_bool_t _enabled = enabled;// 32bit/64bit workaround
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_DISCOVERY_NAPTR(_enabled),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setAoR(const char* ip, int port)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_AOR(ip, port),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setModeServer()
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_MODE_SERVER(),
TSIP_STACK_SET_NULL()) == 0);
}
@ -225,21 +225,21 @@ bool SipStack::setModeServer()
bool SipStack::setSigCompParams(unsigned dms, unsigned sms, unsigned cpb, bool enablePresDict)
{
tsk_bool_t _enablePresDict= enablePresDict;// 32bit/64bit workaround
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_SIGCOMP(dms, sms, cpb, _enablePresDict),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::addSigCompCompartment(const char* compId)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_SIGCOMP_NEW_COMPARTMENT(compId),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::removeSigCompCompartment(const char* compId)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_UNSET_SIGCOMP_COMPARTMENT(compId),
TSIP_STACK_SET_NULL()) == 0);
}
@ -247,14 +247,14 @@ bool SipStack::removeSigCompCompartment(const char* compId)
bool SipStack::setSTUNServer(const char* ip, unsigned short port)
{
unsigned _port = port;//promote
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_STUN_SERVER(ip, _port),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setSTUNCred(const char* login, const char* password)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_STUN_CRED(login, password),
TSIP_STACK_SET_NULL()) == 0);
}
@ -262,14 +262,14 @@ bool SipStack::setSTUNCred(const char* login, const char* password)
bool SipStack::setTLSSecAgree(bool enabled)
{
tsk_bool_t _enable = enabled;
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_SECAGREE_TLS(_enable),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setSSLCretificates(const char* privKey, const char* pubKey, const char* caKey)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_TLS_CERTS(caKey, pubKey, privKey),
TSIP_STACK_SET_NULL()) == 0);
}
@ -277,21 +277,21 @@ bool SipStack::setSSLCretificates(const char* privKey, const char* pubKey, const
bool SipStack::setIPSecSecAgree(bool enabled)
{
tsk_bool_t _enable = enabled;
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_SECAGREE_IPSEC(_enable),
TSIP_STACK_SET_NULL()) == 0);
}
bool SipStack::setIPSecParameters(const char* algo, const char* ealgo, const char* mode, const char* proto)
{
return (tsip_stack_set(this->handle,
return (tsip_stack_set(m_pHandle,
TSIP_STACK_SET_IPSEC_PARAMS(algo, ealgo, mode, proto),
TSIP_STACK_SET_NULL()) == 0);
}
char* SipStack::dnsENUM(const char* service, const char* e164num, const char* domain)
{
tnet_dns_ctx_t* dnsctx = tsip_stack_get_dnsctx(this->handle);
tnet_dns_ctx_t* dnsctx = tsip_stack_get_dnsctx(m_pHandle);
char* uri = tsk_null;
if(dnsctx){
@ -309,7 +309,7 @@ char* SipStack::dnsENUM(const char* service, const char* e164num, const char* do
char* SipStack::dnsNaptrSrv(const char* domain, const char* service, unsigned short *OUTPUT)
{
tnet_dns_ctx_t* dnsctx = tsip_stack_get_dnsctx(this->handle);
tnet_dns_ctx_t* dnsctx = tsip_stack_get_dnsctx(m_pHandle);
char* ip = tsk_null;
tnet_port_t port;
*OUTPUT = 0;
@ -330,7 +330,7 @@ char* SipStack::dnsNaptrSrv(const char* domain, const char* service, unsigned sh
char* SipStack::dnsSrv(const char* service, unsigned short* OUTPUT)
{
tnet_dns_ctx_t* dnsctx = tsip_stack_get_dnsctx(this->handle);
tnet_dns_ctx_t* dnsctx = tsip_stack_get_dnsctx(m_pHandle);
char* ip = tsk_null;
tnet_port_t port = 0;
*OUTPUT = 0;
@ -359,7 +359,7 @@ char* SipStack::getLocalIPnPort(const char* protocol, unsigned short* OUTPUT)
return tsk_null;
}
if((ret = tsip_stack_get_local_ip_n_port(this->handle, protocol, &port, &ip))){
if((ret = tsip_stack_get_local_ip_n_port(m_pHandle, protocol, &port, &ip))){
TSK_DEBUG_ERROR("Failed to get local ip and port with error code=%d", ret);
return tsk_null;
}
@ -370,7 +370,7 @@ char* SipStack::getLocalIPnPort(const char* protocol, unsigned short* OUTPUT)
char* SipStack::getPreferredIdentity()
{
tsip_uri_t* ppid = tsip_stack_get_preferred_id(this->handle);
tsip_uri_t* ppid = tsip_stack_get_preferred_id(m_pHandle);
char* str_ppid = tsk_null;
if(ppid){
str_ppid = tsip_uri_tostring(ppid, tsk_false, tsk_false);
@ -381,12 +381,12 @@ char* SipStack::getPreferredIdentity()
bool SipStack::isValid()
{
return (this->handle != tsk_null);
return (m_pHandle != tsk_null);
}
bool SipStack::stop()
{
int ret = tsip_stack_stop(this->handle);
int ret = tsip_stack_stop(m_pHandle);
return (ret == 0);
}
@ -405,10 +405,10 @@ bool SipStack::isCodecSupported(tdav_codec_id_t codec)
return tdav_codec_is_supported(codec) ? true : false;
}
int stack_callback(const tsip_event_t *sipevent)
static int stack_callback(const tsip_event_t *sipevent)
{
int ret = 0;
const SipStack* Stack = tsk_null;
const SipStack* sipStack = tsk_null;
SipEvent* e = tsk_null;
if(!sipevent){ /* should never happen ...but who know? */
@ -418,89 +418,89 @@ int stack_callback(const tsip_event_t *sipevent)
else {
if(sipevent->type == tsip_event_stack && sipevent->userdata){
/* sessionless event */
Stack = dyn_cast<const SipStack*>((const SipStack*)sipevent->userdata);
sipStack = dyn_cast<const SipStack*>((const SipStack*)sipevent->userdata);
}
else {
const void* userdata;
/* gets the stack from the session */
const tsip_stack_handle_t* stack_handle = tsip_ssession_get_stack(sipevent->ss);
if(stack_handle && (userdata = tsip_stack_get_userdata(stack_handle))){
Stack = dyn_cast<const SipStack*>((const SipStack*)userdata);
sipStack = dyn_cast<const SipStack*>((const SipStack*)userdata);
}
}
}
if(!Stack){
if(!sipStack){
TSK_DEBUG_WARN("Invalid SIP event (Stack is Null).");
return -2;
}
Stack->Lock();
sipStack->Lock();
switch(sipevent->type){
case tsip_event_register:
{ /* REGISTER */
if(Stack->getCallback()){
if(sipStack->getCallback()){
e = new RegistrationEvent(sipevent);
Stack->getCallback()->OnRegistrationEvent((const RegistrationEvent*)e);
sipStack->getCallback()->OnRegistrationEvent((const RegistrationEvent*)e);
}
break;
}
case tsip_event_invite:
{ /* INVITE */
if(Stack->getCallback()){
if(sipStack->getCallback()){
e = new InviteEvent(sipevent);
Stack->getCallback()->OnInviteEvent((const InviteEvent*)e);
sipStack->getCallback()->OnInviteEvent((const InviteEvent*)e);
}
break;
}
case tsip_event_message:
{ /* MESSAGE */
if(Stack->getCallback()){
if(sipStack->getCallback()){
e = new MessagingEvent(sipevent);
Stack->getCallback()->OnMessagingEvent((const MessagingEvent*)e);
sipStack->getCallback()->OnMessagingEvent((const MessagingEvent*)e);
}
break;
}
case tsip_event_options:
{ /* OPTIONS */
if(Stack->getCallback()){
if(sipStack->getCallback()){
e = new OptionsEvent(sipevent);
Stack->getCallback()->OnOptionsEvent((const OptionsEvent*)e);
sipStack->getCallback()->OnOptionsEvent((const OptionsEvent*)e);
}
break;
}
case tsip_event_publish:
{ /* PUBLISH */
if(Stack->getCallback()){
if(sipStack->getCallback()){
e = new PublicationEvent(sipevent);
Stack->getCallback()->OnPublicationEvent((const PublicationEvent*)e);
sipStack->getCallback()->OnPublicationEvent((const PublicationEvent*)e);
}
break;
}
case tsip_event_subscribe:
{ /* SUBSCRIBE */
if(Stack->getCallback()){
if(sipStack->getCallback()){
e = new SubscriptionEvent(sipevent);
Stack->getCallback()->OnSubscriptionEvent((const SubscriptionEvent*)e);
sipStack->getCallback()->OnSubscriptionEvent((const SubscriptionEvent*)e);
}
break;
}
case tsip_event_dialog:
{ /* Common to all dialogs */
if(Stack->getCallback()){
if(sipStack->getCallback()){
e = new DialogEvent(sipevent);
Stack->getCallback()->OnDialogEvent((const DialogEvent*)e);
sipStack->getCallback()->OnDialogEvent((const DialogEvent*)e);
}
break;
}
case tsip_event_stack:
{ /* Stack event */
if(Stack->getCallback()){
if(sipStack->getCallback()){
e = new StackEvent(sipevent);
Stack->getCallback()->OnStackEvent((const StackEvent*)e);
sipStack->getCallback()->OnStackEvent((const StackEvent*)e);
}
break;
}
@ -513,7 +513,7 @@ int stack_callback(const tsip_event_t *sipevent)
}
}
Stack->UnLock();
sipStack->UnLock();
if(e){
delete e;

16
branches/2.0/doubango/bindings/_common/SipStack.h
View File

@ -33,12 +33,12 @@ class DDebugCallback;
class SipStack: public SafeObject
{
public: /* ctor() and dtor() */
SipStack(SipCallback* callback, const char* realm_uri, const char* impi_uri, const char* impu_uri);
SipStack(SipCallback* pCallback, const char* realm_uri, const char* impi_uri, const char* impu_uri);
~SipStack();
public: /* API functions */
bool start();
bool setDebugCallback(DDebugCallback* callback);
bool setDebugCallback(DDebugCallback* pCallback);
bool setRealm(const char* realm_uri);
bool setIMPI(const char* impi);
bool setIMPU(const char* impu_uri);
@ -88,20 +88,20 @@ public: /* API functions */
public: /* Public helper function */
#if !defined(SWIG)
inline tsip_stack_handle_t* getHandle()const{
return this->handle;
return m_pHandle;
}
inline SipCallback* getCallback()const{
return this->callback;
return m_pCallback;
}
inline DDebugCallback* getDebugCallback() const{
return this->debugCallback;
return m_pDebugCallback;
}
#endif
private:
SipCallback* callback;
DDebugCallback* debugCallback;
tsip_stack_handle_t *handle;
SipCallback* m_pCallback;
DDebugCallback* m_pDebugCallback;
tsip_stack_handle_t* m_pHandle;
static unsigned count;
};

424
branches/2.0/doubango/thirdparties/iphone/include/speex/speex.h Normal file
View File

@ -0,0 +1,424 @@
/* Copyright (C) 2002-2006 Jean-Marc Valin*/
/**
@file speex.h
@brief Describes the different modes of the codec
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef SPEEX_H
#define SPEEX_H
/** @defgroup Codec Speex encoder and decoder
* This is the Speex codec itself.
* @{
*/
#include "speex/speex_bits.h"
#include "speex/speex_types.h"
#ifdef __cplusplus
extern "C" {
#endif
/* Values allowed for *ctl() requests */
/** Set enhancement on/off (decoder only) */
#define SPEEX_SET_ENH 0
/** Get enhancement state (decoder only) */
#define SPEEX_GET_ENH 1
/*Would be SPEEX_SET_FRAME_SIZE, but it's (currently) invalid*/
/** Obtain frame size used by encoder/decoder */
#define SPEEX_GET_FRAME_SIZE 3
/** Set quality value */
#define SPEEX_SET_QUALITY 4
/** Get current quality setting */
/* #define SPEEX_GET_QUALITY 5 -- Doesn't make much sense, does it? */
/** Set sub-mode to use */
#define SPEEX_SET_MODE 6
/** Get current sub-mode in use */
#define SPEEX_GET_MODE 7
/** Set low-band sub-mode to use (wideband only)*/
#define SPEEX_SET_LOW_MODE 8
/** Get current low-band mode in use (wideband only)*/
#define SPEEX_GET_LOW_MODE 9
/** Set high-band sub-mode to use (wideband only)*/
#define SPEEX_SET_HIGH_MODE 10
/** Get current high-band mode in use (wideband only)*/
#define SPEEX_GET_HIGH_MODE 11
/** Set VBR on (1) or off (0) */
#define SPEEX_SET_VBR 12
/** Get VBR status (1 for on, 0 for off) */
#define SPEEX_GET_VBR 13
/** Set quality value for VBR encoding (0-10) */
#define SPEEX_SET_VBR_QUALITY 14
/** Get current quality value for VBR encoding (0-10) */
#define SPEEX_GET_VBR_QUALITY 15
/** Set complexity of the encoder (0-10) */
#define SPEEX_SET_COMPLEXITY 16
/** Get current complexity of the encoder (0-10) */
#define SPEEX_GET_COMPLEXITY 17
/** Set bit-rate used by the encoder (or lower) */
#define SPEEX_SET_BITRATE 18
/** Get current bit-rate used by the encoder or decoder */
#define SPEEX_GET_BITRATE 19
/** Define a handler function for in-band Speex request*/
#define SPEEX_SET_HANDLER 20
/** Define a handler function for in-band user-defined request*/
#define SPEEX_SET_USER_HANDLER 22
/** Set sampling rate used in bit-rate computation */
#define SPEEX_SET_SAMPLING_RATE 24
/** Get sampling rate used in bit-rate computation */
#define SPEEX_GET_SAMPLING_RATE 25
/** Reset the encoder/decoder memories to zero*/
#define SPEEX_RESET_STATE 26
/** Get VBR info (mostly used internally) */
#define SPEEX_GET_RELATIVE_QUALITY 29
/** Set VAD status (1 for on, 0 for off) */
#define SPEEX_SET_VAD 30
/** Get VAD status (1 for on, 0 for off) */
#define SPEEX_GET_VAD 31
/** Set Average Bit-Rate (ABR) to n bits per seconds */
#define SPEEX_SET_ABR 32
/** Get Average Bit-Rate (ABR) setting (in bps) */
#define SPEEX_GET_ABR 33
/** Set DTX status (1 for on, 0 for off) */
#define SPEEX_SET_DTX 34
/** Get DTX status (1 for on, 0 for off) */
#define SPEEX_GET_DTX 35
/** Set submode encoding in each frame (1 for yes, 0 for no, setting to no breaks the standard) */
#define SPEEX_SET_SUBMODE_ENCODING 36
/** Get submode encoding in each frame */
#define SPEEX_GET_SUBMODE_ENCODING 37
/*#define SPEEX_SET_LOOKAHEAD 38*/
/** Returns the lookahead used by Speex */
#define SPEEX_GET_LOOKAHEAD 39
/** Sets tuning for packet-loss concealment (expected loss rate) */
#define SPEEX_SET_PLC_TUNING 40
/** Gets tuning for PLC */
#define SPEEX_GET_PLC_TUNING 41
/** Sets the max bit-rate allowed in VBR mode */
#define SPEEX_SET_VBR_MAX_BITRATE 42
/** Gets the max bit-rate allowed in VBR mode */
#define SPEEX_GET_VBR_MAX_BITRATE 43
/** Turn on/off input/output high-pass filtering */
#define SPEEX_SET_HIGHPASS 44
/** Get status of input/output high-pass filtering */
#define SPEEX_GET_HIGHPASS 45
/** Get "activity level" of the last decoded frame, i.e.
how much damage we cause if we remove the frame */
#define SPEEX_GET_ACTIVITY 47
/* Preserving compatibility:*/
/** Equivalent to SPEEX_SET_ENH */
#define SPEEX_SET_PF 0
/** Equivalent to SPEEX_GET_ENH */
#define SPEEX_GET_PF 1
/* Values allowed for mode queries */
/** Query the frame size of a mode */
#define SPEEX_MODE_FRAME_SIZE 0
/** Query the size of an encoded frame for a particular sub-mode */
#define SPEEX_SUBMODE_BITS_PER_FRAME 1
/** Get major Speex version */
#define SPEEX_LIB_GET_MAJOR_VERSION 1
/** Get minor Speex version */
#define SPEEX_LIB_GET_MINOR_VERSION 3
/** Get micro Speex version */
#define SPEEX_LIB_GET_MICRO_VERSION 5
/** Get extra Speex version */
#define SPEEX_LIB_GET_EXTRA_VERSION 7
/** Get Speex version string */
#define SPEEX_LIB_GET_VERSION_STRING 9
/*#define SPEEX_LIB_SET_ALLOC_FUNC 10
#define SPEEX_LIB_GET_ALLOC_FUNC 11
#define SPEEX_LIB_SET_FREE_FUNC 12
#define SPEEX_LIB_GET_FREE_FUNC 13
#define SPEEX_LIB_SET_WARNING_FUNC 14
#define SPEEX_LIB_GET_WARNING_FUNC 15
#define SPEEX_LIB_SET_ERROR_FUNC 16
#define SPEEX_LIB_GET_ERROR_FUNC 17
*/
/** Number of defined modes in Speex */
#define SPEEX_NB_MODES 3
/** modeID for the defined narrowband mode */
#define SPEEX_MODEID_NB 0
/** modeID for the defined wideband mode */
#define SPEEX_MODEID_WB 1
/** modeID for the defined ultra-wideband mode */
#define SPEEX_MODEID_UWB 2
struct SpeexMode;
/* Prototypes for mode function pointers */
/** Encoder state initialization function */
typedef void *(*encoder_init_func)(const struct SpeexMode *mode);
/** Encoder state destruction function */
typedef void (*encoder_destroy_func)(void *st);
/** Main encoding function */
typedef int (*encode_func)(void *state, void *in, SpeexBits *bits);
/** Function for controlling the encoder options */
typedef int (*encoder_ctl_func)(void *state, int request, void *ptr);
/** Decoder state initialization function */
typedef void *(*decoder_init_func)(const struct SpeexMode *mode);
/** Decoder state destruction function */
typedef void (*decoder_destroy_func)(void *st);
/** Main decoding function */
typedef int (*decode_func)(void *state, SpeexBits *bits, void *out);
/** Function for controlling the decoder options */
typedef int (*decoder_ctl_func)(void *state, int request, void *ptr);
/** Query function for a mode */
typedef int (*mode_query_func)(const void *mode, int request, void *ptr);
/** Struct defining a Speex mode */
typedef struct SpeexMode {
/** Pointer to the low-level mode data */
const void *mode;
/** Pointer to the mode query function */
mode_query_func query;
/** The name of the mode (you should not rely on this to identify the mode)*/
const char *modeName;
/**ID of the mode*/
int modeID;
/**Version number of the bitstream (incremented every time we break
bitstream compatibility*/
int bitstream_version;
/** Pointer to encoder initialization function */
encoder_init_func enc_init;
/** Pointer to encoder destruction function */
encoder_destroy_func enc_destroy;
/** Pointer to frame encoding function */
encode_func enc;
/** Pointer to decoder initialization function */
decoder_init_func dec_init;
/** Pointer to decoder destruction function */
decoder_destroy_func dec_destroy;
/** Pointer to frame decoding function */
decode_func dec;
/** ioctl-like requests for encoder */
encoder_ctl_func enc_ctl;
/** ioctl-like requests for decoder */
decoder_ctl_func dec_ctl;
} SpeexMode;
/**
* Returns a handle to a newly created Speex encoder state structure. For now,
* the "mode" argument can be &nb_mode or &wb_mode . In the future, more modes
* may be added. Note that for now if you have more than one channels to
* encode, you need one state per channel.
*
* @param mode The mode to use (either speex_nb_mode or speex_wb.mode)
* @return A newly created encoder state or NULL if state allocation fails
*/
void *speex_encoder_init(const SpeexMode *mode);
/** Frees all resources associated to an existing Speex encoder state.
* @param state Encoder state to be destroyed */
void speex_encoder_destroy(void *state);
/** Uses an existing encoder state to encode one frame of speech pointed to by
"in". The encoded bit-stream is saved in "bits".
@param state Encoder state
@param in Frame that will be encoded with a +-2^15 range. This data MAY be
overwritten by the encoder and should be considered uninitialised
after the call.
@param bits Bit-stream where the data will be written
@return 0 if frame needs not be transmitted (DTX only), 1 otherwise
*/
int speex_encode(void *state, float *in, SpeexBits *bits);
/** Uses an existing encoder state to encode one frame of speech pointed to by
"in". The encoded bit-stream is saved in "bits".
@param state Encoder state
@param in Frame that will be encoded with a +-2^15 range
@param bits Bit-stream where the data will be written
@return 0 if frame needs not be transmitted (DTX only), 1 otherwise
*/
int speex_encode_int(void *state, spx_int16_t *in, SpeexBits *bits);
/** Used like the ioctl function to control the encoder parameters
*
* @param state Encoder state
* @param request ioctl-type request (one of the SPEEX_* macros)
* @param ptr Data exchanged to-from function
* @return 0 if no error, -1 if request in unknown, -2 for invalid parameter
*/
int speex_encoder_ctl(void *state, int request, void *ptr);
/** Returns a handle to a newly created decoder state structure. For now,
* the mode argument can be &nb_mode or &wb_mode . In the future, more modes
* may be added. Note that for now if you have more than one channels to
* decode, you need one state per channel.
*
* @param mode Speex mode (one of speex_nb_mode or speex_wb_mode)
* @return A newly created decoder state or NULL if state allocation fails
*/
void *speex_decoder_init(const SpeexMode *mode);
/** Frees all resources associated to an existing decoder state.
*
* @param state State to be destroyed
*/
void speex_decoder_destroy(void *state);
/** Uses an existing decoder state to decode one frame of speech from
* bit-stream bits. The output speech is saved written to out.
*
* @param state Decoder state
* @param bits Bit-stream from which to decode the frame (NULL if the packet was lost)
* @param out Where to write the decoded frame
* @return return status (0 for no error, -1 for end of stream, -2 corrupt stream)
*/
int speex_decode(void *state, SpeexBits *bits, float *out);
/** Uses an existing decoder state to decode one frame of speech from
* bit-stream bits. The output speech is saved written to out.
*
* @param state Decoder state
* @param bits Bit-stream from which to decode the frame (NULL if the packet was lost)
* @param out Where to write the decoded frame
* @return return status (0 for no error, -1 for end of stream, -2 corrupt stream)
*/
int speex_decode_int(void *state, SpeexBits *bits, spx_int16_t *out);
/** Used like the ioctl function to control the encoder parameters
*
* @param state Decoder state
* @param request ioctl-type request (one of the SPEEX_* macros)
* @param ptr Data exchanged to-from function
* @return 0 if no error, -1 if request in unknown, -2 for invalid parameter
*/
int speex_decoder_ctl(void *state, int request, void *ptr);
/** Query function for mode information
*
* @param mode Speex mode
* @param request ioctl-type request (one of the SPEEX_* macros)
* @param ptr Data exchanged to-from function
* @return 0 if no error, -1 if request in unknown, -2 for invalid parameter
*/
int speex_mode_query(const SpeexMode *mode, int request, void *ptr);
/** Functions for controlling the behavior of libspeex
* @param request ioctl-type request (one of the SPEEX_LIB_* macros)
* @param ptr Data exchanged to-from function
* @return 0 if no error, -1 if request in unknown, -2 for invalid parameter
*/
int speex_lib_ctl(int request, void *ptr);
/** Default narrowband mode */
extern const SpeexMode speex_nb_mode;
/** Default wideband mode */
extern const SpeexMode speex_wb_mode;
/** Default "ultra-wideband" mode */
extern const SpeexMode speex_uwb_mode;
/** List of all modes available */
extern const SpeexMode * const speex_mode_list[SPEEX_NB_MODES];
/** Obtain one of the modes available */
const SpeexMode * speex_lib_get_mode (int mode);
#ifndef WIN32
/* We actually override the function in the narrowband case so that we can avoid linking in the wideband stuff */
#define speex_lib_get_mode(mode) ((mode)==SPEEX_MODEID_NB ? &speex_nb_mode : speex_lib_get_mode (mode))
#endif
#ifdef __cplusplus
}
#endif
/** @}*/
#endif

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/* Copyright (C) 2002 Jean-Marc Valin */
/**
@file speex_bits.h
@brief Handles bit packing/unpacking
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef BITS_H
#define BITS_H
/** @defgroup SpeexBits SpeexBits: Bit-stream manipulations
* This is the structure that holds the bit-stream when encoding or decoding
* with Speex. It allows some manipulations as well.
* @{
*/
#ifdef __cplusplus
extern "C" {
#endif
/** Bit-packing data structure representing (part of) a bit-stream. */
typedef struct SpeexBits {
char *chars; /**< "raw" data */
int nbBits; /**< Total number of bits stored in the stream*/
int charPtr; /**< Position of the byte "cursor" */
int bitPtr; /**< Position of the bit "cursor" within the current char */
int owner; /**< Does the struct "own" the "raw" buffer (member "chars") */
int overflow;/**< Set to one if we try to read past the valid data */
int buf_size;/**< Allocated size for buffer */
int reserved1; /**< Reserved for future use */
void *reserved2; /**< Reserved for future use */
} SpeexBits;
/** Initializes and allocates resources for a SpeexBits struct */
void speex_bits_init(SpeexBits *bits);
/** Initializes SpeexBits struct using a pre-allocated buffer*/
void speex_bits_init_buffer(SpeexBits *bits, void *buff, int buf_size);
/** Sets the bits in a SpeexBits struct to use data from an existing buffer (for decoding without copying data) */
void speex_bits_set_bit_buffer(SpeexBits *bits, void *buff, int buf_size);
/** Frees all resources associated to a SpeexBits struct. Right now this does nothing since no resources are allocated, but this could change in the future.*/
void speex_bits_destroy(SpeexBits *bits);
/** Resets bits to initial value (just after initialization, erasing content)*/
void speex_bits_reset(SpeexBits *bits);
/** Rewind the bit-stream to the beginning (ready for read) without erasing the content */
void speex_bits_rewind(SpeexBits *bits);
/** Initializes the bit-stream from the data in an area of memory */
void speex_bits_read_from(SpeexBits *bits, char *bytes, int len);
/** Append bytes to the bit-stream
*
* @param bits Bit-stream to operate on
* @param bytes pointer to the bytes what will be appended
* @param len Number of bytes of append
*/
void speex_bits_read_whole_bytes(SpeexBits *bits, char *bytes, int len);
/** Write the content of a bit-stream to an area of memory
*
* @param bits Bit-stream to operate on
* @param bytes Memory location where to write the bits
* @param max_len Maximum number of bytes to write (i.e. size of the "bytes" buffer)
* @return Number of bytes written to the "bytes" buffer
*/
int speex_bits_write(SpeexBits *bits, char *bytes, int max_len);
/** Like speex_bits_write, but writes only the complete bytes in the stream. Also removes the written bytes from the stream */
int speex_bits_write_whole_bytes(SpeexBits *bits, char *bytes, int max_len);
/** Append bits to the bit-stream
* @param bits Bit-stream to operate on
* @param data Value to append as integer
* @param nbBits number of bits to consider in "data"
*/
void speex_bits_pack(SpeexBits *bits, int data, int nbBits);
/** Interpret the next bits in the bit-stream as a signed integer
*
* @param bits Bit-stream to operate on
* @param nbBits Number of bits to interpret
* @return A signed integer represented by the bits read
*/
int speex_bits_unpack_signed(SpeexBits *bits, int nbBits);
/** Interpret the next bits in the bit-stream as an unsigned integer
*
* @param bits Bit-stream to operate on
* @param nbBits Number of bits to interpret
* @return An unsigned integer represented by the bits read
*/
unsigned int speex_bits_unpack_unsigned(SpeexBits *bits, int nbBits);
/** Returns the number of bytes in the bit-stream, including the last one even if it is not "full"
*
* @param bits Bit-stream to operate on
* @return Number of bytes in the stream
*/
int speex_bits_nbytes(SpeexBits *bits);
/** Same as speex_bits_unpack_unsigned, but without modifying the cursor position
*
* @param bits Bit-stream to operate on
* @param nbBits Number of bits to look for
* @return Value of the bits peeked, interpreted as unsigned
*/
unsigned int speex_bits_peek_unsigned(SpeexBits *bits, int nbBits);
/** Get the value of the next bit in the stream, without modifying the
* "cursor" position
*
* @param bits Bit-stream to operate on
* @return Value of the bit peeked (one bit only)
*/
int speex_bits_peek(SpeexBits *bits);
/** Advances the position of the "bit cursor" in the stream
*
* @param bits Bit-stream to operate on
* @param n Number of bits to advance
*/
void speex_bits_advance(SpeexBits *bits, int n);
/** Returns the number of bits remaining to be read in a stream
*
* @param bits Bit-stream to operate on
* @return Number of bits that can still be read from the stream
*/
int speex_bits_remaining(SpeexBits *bits);
/** Insert a terminator so that the data can be sent as a packet while auto-detecting
* the number of frames in each packet
*
* @param bits Bit-stream to operate on
*/
void speex_bits_insert_terminator(SpeexBits *bits);
#ifdef __cplusplus
}
#endif
/* @} */
#endif

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/* Copyright (C) 2007 Jean-Marc Valin
File: speex_buffer.h
This is a very simple ring buffer implementation. It is not thread-safe
so you need to do your own locking.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:
1. Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
3. The name of the author may not be used to endorse or promote products
derived from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef SPEEX_BUFFER_H
#define SPEEX_BUFFER_H
#include "speex/speex_types.h"
#ifdef __cplusplus
extern "C" {
#endif
struct SpeexBuffer_;
typedef struct SpeexBuffer_ SpeexBuffer;
SpeexBuffer *speex_buffer_init(int size);
void speex_buffer_destroy(SpeexBuffer *st);
int speex_buffer_write(SpeexBuffer *st, void *data, int len);
int speex_buffer_writezeros(SpeexBuffer *st, int len);
int speex_buffer_read(SpeexBuffer *st, void *data, int len);
int speex_buffer_get_available(SpeexBuffer *st);
int speex_buffer_resize(SpeexBuffer *st, int len);
#ifdef __cplusplus
}
#endif
#endif

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/* Copyright (C) 2002 Jean-Marc Valin*/
/**
@file speex_callbacks.h
@brief Describes callback handling and in-band signalling
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef SPEEX_CALLBACKS_H
#define SPEEX_CALLBACKS_H
/** @defgroup SpeexCallbacks Various definitions for Speex callbacks supported by the decoder.
* @{
*/
#include "speex.h"
#ifdef __cplusplus
extern "C" {
#endif
/** Total number of callbacks */
#define SPEEX_MAX_CALLBACKS 16
/* Describes all the in-band requests */
/*These are 1-bit requests*/
/** Request for perceptual enhancement (1 for on, 0 for off) */
#define SPEEX_INBAND_ENH_REQUEST 0
/** Reserved */
#define SPEEX_INBAND_RESERVED1 1
/*These are 4-bit requests*/
/** Request for a mode change */
#define SPEEX_INBAND_MODE_REQUEST 2
/** Request for a low mode change */
#define SPEEX_INBAND_LOW_MODE_REQUEST 3
/** Request for a high mode change */
#define SPEEX_INBAND_HIGH_MODE_REQUEST 4
/** Request for VBR (1 on, 0 off) */
#define SPEEX_INBAND_VBR_QUALITY_REQUEST 5
/** Request to be sent acknowledge */
#define SPEEX_INBAND_ACKNOWLEDGE_REQUEST 6
/** Request for VBR (1 for on, 0 for off) */
#define SPEEX_INBAND_VBR_REQUEST 7
/*These are 8-bit requests*/
/** Send a character in-band */
#define SPEEX_INBAND_CHAR 8
/** Intensity stereo information */
#define SPEEX_INBAND_STEREO 9
/*These are 16-bit requests*/
/** Transmit max bit-rate allowed */
#define SPEEX_INBAND_MAX_BITRATE 10
/*These are 32-bit requests*/
/** Acknowledge packet reception */
#define SPEEX_INBAND_ACKNOWLEDGE 12
/** Callback function type */
typedef int (*speex_callback_func)(SpeexBits *bits, void *state, void *data);
/** Callback information */
typedef struct SpeexCallback {
int callback_id; /**< ID associated to the callback */
speex_callback_func func; /**< Callback handler function */
void *data; /**< Data that will be sent to the handler */
void *reserved1; /**< Reserved for future use */
int reserved2; /**< Reserved for future use */
} SpeexCallback;
/** Handle in-band request */
int speex_inband_handler(SpeexBits *bits, SpeexCallback *callback_list, void *state);
/** Standard handler for mode request (change mode, no questions asked) */
int speex_std_mode_request_handler(SpeexBits *bits, void *state, void *data);
/** Standard handler for high mode request (change high mode, no questions asked) */
int speex_std_high_mode_request_handler(SpeexBits *bits, void *state, void *data);
/** Standard handler for in-band characters (write to stderr) */
int speex_std_char_handler(SpeexBits *bits, void *state, void *data);
/** Default handler for user-defined requests: in this case, just ignore */
int speex_default_user_handler(SpeexBits *bits, void *state, void *data);
/** Standard handler for low mode request (change low mode, no questions asked) */
int speex_std_low_mode_request_handler(SpeexBits *bits, void *state, void *data);
/** Standard handler for VBR request (Set VBR, no questions asked) */
int speex_std_vbr_request_handler(SpeexBits *bits, void *state, void *data);
/** Standard handler for enhancer request (Turn enhancer on/off, no questions asked) */
int speex_std_enh_request_handler(SpeexBits *bits, void *state, void *data);
/** Standard handler for VBR quality request (Set VBR quality, no questions asked) */
int speex_std_vbr_quality_request_handler(SpeexBits *bits, void *state, void *data);
#ifdef __cplusplus
}
#endif
/** @} */
#endif

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/* Copyright (C) Jean-Marc Valin */
/**
@file speex_echo.h
@brief Echo cancellation
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:
1. Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
3. The name of the author may not be used to endorse or promote products
derived from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef SPEEX_ECHO_H
#define SPEEX_ECHO_H
/** @defgroup SpeexEchoState SpeexEchoState: Acoustic echo canceller
* This is the acoustic echo canceller module.
* @{
*/
#include "speex/speex_types.h"
#ifdef __cplusplus
extern "C" {
#endif
/** Obtain frame size used by the AEC */
#define SPEEX_ECHO_GET_FRAME_SIZE 3
/** Set sampling rate */
#define SPEEX_ECHO_SET_SAMPLING_RATE 24
/** Get sampling rate */
#define SPEEX_ECHO_GET_SAMPLING_RATE 25
/* Can't set window sizes */
/** Get size of impulse response (int32) */
#define SPEEX_ECHO_GET_IMPULSE_RESPONSE_SIZE 27
/* Can't set window content */
/** Get impulse response (int32[]) */
#define SPEEX_ECHO_GET_IMPULSE_RESPONSE 29
/** Internal echo canceller state. Should never be accessed directly. */
struct SpeexEchoState_;
/** @class SpeexEchoState
* This holds the state of the echo canceller. You need one per channel.
*/
/** Internal echo canceller state. Should never be accessed directly. */
typedef struct SpeexEchoState_ SpeexEchoState;
/** Creates a new echo canceller state
* @param frame_size Number of samples to process at one time (should correspond to 10-20 ms)
* @param filter_length Number of samples of echo to cancel (should generally correspond to 100-500 ms)
* @return Newly-created echo canceller state
*/
SpeexEchoState *speex_echo_state_init(int frame_size, int filter_length);
/** Creates a new multi-channel echo canceller state
* @param frame_size Number of samples to process at one time (should correspond to 10-20 ms)
* @param filter_length Number of samples of echo to cancel (should generally correspond to 100-500 ms)
* @param nb_mic Number of microphone channels
* @param nb_speakers Number of speaker channels
* @return Newly-created echo canceller state
*/
SpeexEchoState *speex_echo_state_init_mc(int frame_size, int filter_length, int nb_mic, int nb_speakers);
/** Destroys an echo canceller state
* @param st Echo canceller state
*/
void speex_echo_state_destroy(SpeexEchoState *st);
/** Performs echo cancellation a frame, based on the audio sent to the speaker (no delay is added
* to playback in this form)
*
* @param st Echo canceller state
* @param rec Signal from the microphone (near end + far end echo)
* @param play Signal played to the speaker (received from far end)
* @param out Returns near-end signal with echo removed
*/
void speex_echo_cancellation(SpeexEchoState *st, const spx_int16_t *rec, const spx_int16_t *play, spx_int16_t *out);
/** Performs echo cancellation a frame (deprecated) */
void speex_echo_cancel(SpeexEchoState *st, const spx_int16_t *rec, const spx_int16_t *play, spx_int16_t *out, spx_int32_t *Yout);
/** Perform echo cancellation using internal playback buffer, which is delayed by two frames
* to account for the delay introduced by most soundcards (but it could be off!)
* @param st Echo canceller state
* @param rec Signal from the microphone (near end + far end echo)
* @param out Returns near-end signal with echo removed
*/
void speex_echo_capture(SpeexEchoState *st, const spx_int16_t *rec, spx_int16_t *out);
/** Let the echo canceller know that a frame was just queued to the soundcard
* @param st Echo canceller state
* @param play Signal played to the speaker (received from far end)
*/
void speex_echo_playback(SpeexEchoState *st, const spx_int16_t *play);
/** Reset the echo canceller to its original state
* @param st Echo canceller state
*/
void speex_echo_state_reset(SpeexEchoState *st);
/** Used like the ioctl function to control the echo canceller parameters
*
* @param st Echo canceller state
* @param request ioctl-type request (one of the SPEEX_ECHO_* macros)
* @param ptr Data exchanged to-from function
* @return 0 if no error, -1 if request in unknown
*/
int speex_echo_ctl(SpeexEchoState *st, int request, void *ptr);
struct SpeexDecorrState_;
typedef struct SpeexDecorrState_ SpeexDecorrState;
/** Create a state for the channel decorrelation algorithm
This is useful for multi-channel echo cancellation only
* @param rate Sampling rate
* @param channels Number of channels (it's a bit pointless if you don't have at least 2)
* @param frame_size Size of the frame to process at ones (counting samples *per* channel)
*/
SpeexDecorrState *speex_decorrelate_new(int rate, int channels, int frame_size);
/** Remove correlation between the channels by modifying the phase and possibly
adding noise in a way that is not (or little) perceptible.
* @param st Decorrelator state
* @param in Input audio in interleaved format
* @param out Result of the decorrelation (out *may* alias in)
* @param strength How much alteration of the audio to apply from 0 to 100.
*/
void speex_decorrelate(SpeexDecorrState *st, const spx_int16_t *in, spx_int16_t *out, int strength);
/** Destroy a Decorrelation state
* @param st State to destroy
*/
void speex_decorrelate_destroy(SpeexDecorrState *st);
#ifdef __cplusplus
}
#endif
/** @}*/
#endif

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/* Copyright (C) 2002 Jean-Marc Valin */
/**
@file speex_header.h
@brief Describes the Speex header
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef SPEEX_HEADER_H
#define SPEEX_HEADER_H
/** @defgroup SpeexHeader SpeexHeader: Makes it easy to write/parse an Ogg/Speex header
* This is the Speex header for the Ogg encapsulation. You don't need that if you just use RTP.
* @{
*/
#include "speex/speex_types.h"
#ifdef __cplusplus
extern "C" {
#endif
struct SpeexMode;
/** Length of the Speex header identifier */
#define SPEEX_HEADER_STRING_LENGTH 8
/** Maximum number of characters for encoding the Speex version number in the header */
#define SPEEX_HEADER_VERSION_LENGTH 20
/** Speex header info for file-based formats */
typedef struct SpeexHeader {
char speex_string[SPEEX_HEADER_STRING_LENGTH]; /**< Identifies a Speex bit-stream, always set to "Speex " */
char speex_version[SPEEX_HEADER_VERSION_LENGTH]; /**< Speex version */
spx_int32_t speex_version_id; /**< Version for Speex (for checking compatibility) */
spx_int32_t header_size; /**< Total size of the header ( sizeof(SpeexHeader) ) */
spx_int32_t rate; /**< Sampling rate used */
spx_int32_t mode; /**< Mode used (0 for narrowband, 1 for wideband) */
spx_int32_t mode_bitstream_version; /**< Version ID of the bit-stream */
spx_int32_t nb_channels; /**< Number of channels encoded */
spx_int32_t bitrate; /**< Bit-rate used */
spx_int32_t frame_size; /**< Size of frames */
spx_int32_t vbr; /**< 1 for a VBR encoding, 0 otherwise */
spx_int32_t frames_per_packet; /**< Number of frames stored per Ogg packet */
spx_int32_t extra_headers; /**< Number of additional headers after the comments */
spx_int32_t reserved1; /**< Reserved for future use, must be zero */
spx_int32_t reserved2; /**< Reserved for future use, must be zero */
} SpeexHeader;
/** Initializes a SpeexHeader using basic information */
void speex_init_header(SpeexHeader *header, int rate, int nb_channels, const struct SpeexMode *m);
/** Creates the header packet from the header itself (mostly involves endianness conversion) */
char *speex_header_to_packet(SpeexHeader *header, int *size);
/** Creates a SpeexHeader from a packet */
SpeexHeader *speex_packet_to_header(char *packet, int size);
/** Frees the memory allocated by either speex_header_to_packet() or speex_packet_to_header() */
void speex_header_free(void *ptr);
#ifdef __cplusplus
}
#endif
/** @} */
#endif

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/* Copyright (C) 2002 Jean-Marc Valin */
/**
@file speex_jitter.h
@brief Adaptive jitter buffer for Speex
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef SPEEX_JITTER_H
#define SPEEX_JITTER_H
/** @defgroup JitterBuffer JitterBuffer: Adaptive jitter buffer
* This is the jitter buffer that reorders UDP/RTP packets and adjusts the buffer size
* to maintain good quality and low latency.
* @{
*/
#include "speex/speex_types.h"
#ifdef __cplusplus
extern "C" {
#endif
/** Generic adaptive jitter buffer state */
struct JitterBuffer_;
/** Generic adaptive jitter buffer state */
typedef struct JitterBuffer_ JitterBuffer;
/** Definition of an incoming packet */
typedef struct _JitterBufferPacket JitterBufferPacket;
/** Definition of an incoming packet */
struct _JitterBufferPacket {
char *data; /**< Data bytes contained in the packet */
spx_uint32_t len; /**< Length of the packet in bytes */
spx_uint32_t timestamp; /**< Timestamp for the packet */
spx_uint32_t span; /**< Time covered by the packet (same units as timestamp) */
spx_uint16_t sequence; /**< RTP Sequence number if available (0 otherwise) */
spx_uint32_t user_data; /**< Put whatever data you like here (it's ignored by the jitter buffer) */
};
/** Packet has been retrieved */
#define JITTER_BUFFER_OK 0
/** Packet is lost or is late */
#define JITTER_BUFFER_MISSING 1
/** A "fake" packet is meant to be inserted here to increase buffering */
#define JITTER_BUFFER_INSERTION 2
/** There was an error in the jitter buffer */
#define JITTER_BUFFER_INTERNAL_ERROR -1
/** Invalid argument */
#define JITTER_BUFFER_BAD_ARGUMENT -2
/** Set minimum amount of extra buffering required (margin) */
#define JITTER_BUFFER_SET_MARGIN 0
/** Get minimum amount of extra buffering required (margin) */
#define JITTER_BUFFER_GET_MARGIN 1
/* JITTER_BUFFER_SET_AVAILABLE_COUNT wouldn't make sense */
/** Get the amount of available packets currently buffered */
#define JITTER_BUFFER_GET_AVAILABLE_COUNT 3
/** Included because of an early misspelling (will remove in next release) */
#define JITTER_BUFFER_GET_AVALIABLE_COUNT 3
/** Assign a function to destroy unused packet. When setting that, the jitter
buffer no longer copies packet data. */
#define JITTER_BUFFER_SET_DESTROY_CALLBACK 4
/** */
#define JITTER_BUFFER_GET_DESTROY_CALLBACK 5
/** Tell the jitter buffer to only adjust the delay in multiples of the step parameter provided */
#define JITTER_BUFFER_SET_DELAY_STEP 6
/** */
#define JITTER_BUFFER_GET_DELAY_STEP 7
/** Tell the jitter buffer to only do concealment in multiples of the size parameter provided */
#define JITTER_BUFFER_SET_CONCEALMENT_SIZE 8
#define JITTER_BUFFER_GET_CONCEALMENT_SIZE 9
/** Absolute max amount of loss that can be tolerated regardless of the delay. Typical loss
should be half of that or less. */
#define JITTER_BUFFER_SET_MAX_LATE_RATE 10
#define JITTER_BUFFER_GET_MAX_LATE_RATE 11
/** Equivalent cost of one percent late packet in timestamp units */
#define JITTER_BUFFER_SET_LATE_COST 12
#define JITTER_BUFFER_GET_LATE_COST 13
/** Initialises jitter buffer
*
* @param step_size Starting value for the size of concleanment packets and delay
adjustment steps. Can be changed at any time using JITTER_BUFFER_SET_DELAY_STEP
and JITTER_BUFFER_GET_CONCEALMENT_SIZE.
* @return Newly created jitter buffer state
*/
JitterBuffer *jitter_buffer_init(int step_size);
/** Restores jitter buffer to its original state
*
* @param jitter Jitter buffer state
*/
void jitter_buffer_reset(JitterBuffer *jitter);
/** Destroys jitter buffer
*
* @param jitter Jitter buffer state
*/
void jitter_buffer_destroy(JitterBuffer *jitter);
/** Put one packet into the jitter buffer
*
* @param jitter Jitter buffer state
* @param packet Incoming packet
*/
void jitter_buffer_put(JitterBuffer *jitter, const JitterBufferPacket *packet);
/** Get one packet from the jitter buffer
*
* @param jitter Jitter buffer state
* @param packet Returned packet
* @param desired_span Number of samples (or units) we wish to get from the buffer (no guarantee)
* @param current_timestamp Timestamp for the returned packet
*/
int jitter_buffer_get(JitterBuffer *jitter, JitterBufferPacket *packet, spx_int32_t desired_span, spx_int32_t *start_offset);
/** Used right after jitter_buffer_get() to obtain another packet that would have the same timestamp.
* This is mainly useful for media where a single "frame" can be split into several packets.
*
* @param jitter Jitter buffer state
* @param packet Returned packet
*/
int jitter_buffer_get_another(JitterBuffer *jitter, JitterBufferPacket *packet);
/** Get pointer timestamp of jitter buffer
*
* @param jitter Jitter buffer state
*/
int jitter_buffer_get_pointer_timestamp(JitterBuffer *jitter);
/** Advance by one tick
*
* @param jitter Jitter buffer state
*/
void jitter_buffer_tick(JitterBuffer *jitter);
/** Telling the jitter buffer about the remaining data in the application buffer
* @param jitter Jitter buffer state
* @param rem Amount of data buffered by the application (timestamp units)
*/
void jitter_buffer_remaining_span(JitterBuffer *jitter, spx_uint32_t rem);
/** Used like the ioctl function to control the jitter buffer parameters
*
* @param jitter Jitter buffer state
* @param request ioctl-type request (one of the JITTER_BUFFER_* macros)
* @param ptr Data exchanged to-from function
* @return 0 if no error, -1 if request in unknown
*/
int jitter_buffer_ctl(JitterBuffer *jitter, int request, void *ptr);
int jitter_buffer_update_delay(JitterBuffer *jitter, JitterBufferPacket *packet, spx_int32_t *start_offset);
/* @} */
#ifdef __cplusplus
}
#endif
#endif

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/* Copyright (C) 2003 Epic Games
Written by Jean-Marc Valin */
/**
* @file speex_preprocess.h
* @brief Speex preprocessor. The preprocess can do noise suppression,
* residual echo suppression (after using the echo canceller), automatic
* gain control (AGC) and voice activity detection (VAD).
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:
1. Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
3. The name of the author may not be used to endorse or promote products
derived from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef SPEEX_PREPROCESS_H
#define SPEEX_PREPROCESS_H
/** @defgroup SpeexPreprocessState SpeexPreprocessState: The Speex preprocessor
* This is the Speex preprocessor. The preprocess can do noise suppression,
* residual echo suppression (after using the echo canceller), automatic
* gain control (AGC) and voice activity detection (VAD).
* @{
*/
#include "speex/speex_types.h"
#ifdef __cplusplus
extern "C" {
#endif
/** State of the preprocessor (one per channel). Should never be accessed directly. */
struct SpeexPreprocessState_;
/** State of the preprocessor (one per channel). Should never be accessed directly. */
typedef struct SpeexPreprocessState_ SpeexPreprocessState;
/** Creates a new preprocessing state. You MUST create one state per channel processed.
* @param frame_size Number of samples to process at one time (should correspond to 10-20 ms). Must be
* the same value as that used for the echo canceller for residual echo cancellation to work.
* @param sampling_rate Sampling rate used for the input.
* @return Newly created preprocessor state
*/
SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_rate);
/** Destroys a preprocessor state
* @param st Preprocessor state to destroy
*/
void speex_preprocess_state_destroy(SpeexPreprocessState *st);
/** Preprocess a frame
* @param st Preprocessor state
* @param x Audio sample vector (in and out). Must be same size as specified in speex_preprocess_state_init().
* @return Bool value for voice activity (1 for speech, 0 for noise/silence), ONLY if VAD turned on.
*/
int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x);
/** Preprocess a frame (deprecated, use speex_preprocess_run() instead)*/
int speex_preprocess(SpeexPreprocessState *st, spx_int16_t *x, spx_int32_t *echo);
/** Update preprocessor state, but do not compute the output
* @param st Preprocessor state
* @param x Audio sample vector (in only). Must be same size as specified in speex_preprocess_state_init().
*/
void speex_preprocess_estimate_update(SpeexPreprocessState *st, spx_int16_t *x);
/** Used like the ioctl function to control the preprocessor parameters
* @param st Preprocessor state
* @param request ioctl-type request (one of the SPEEX_PREPROCESS_* macros)
* @param ptr Data exchanged to-from function
* @return 0 if no error, -1 if request in unknown
*/
int speex_preprocess_ctl(SpeexPreprocessState *st, int request, void *ptr);
/** Set preprocessor denoiser state */
#define SPEEX_PREPROCESS_SET_DENOISE 0
/** Get preprocessor denoiser state */
#define SPEEX_PREPROCESS_GET_DENOISE 1
/** Set preprocessor Automatic Gain Control state */
#define SPEEX_PREPROCESS_SET_AGC 2
/** Get preprocessor Automatic Gain Control state */
#define SPEEX_PREPROCESS_GET_AGC 3
/** Set preprocessor Voice Activity Detection state */
#define SPEEX_PREPROCESS_SET_VAD 4
/** Get preprocessor Voice Activity Detection state */
#define SPEEX_PREPROCESS_GET_VAD 5
/** Set preprocessor Automatic Gain Control level (float) */
#define SPEEX_PREPROCESS_SET_AGC_LEVEL 6
/** Get preprocessor Automatic Gain Control level (float) */
#define SPEEX_PREPROCESS_GET_AGC_LEVEL 7
/** Set preprocessor dereverb state */
#define SPEEX_PREPROCESS_SET_DEREVERB 8
/** Get preprocessor dereverb state */
#define SPEEX_PREPROCESS_GET_DEREVERB 9
/** Set preprocessor dereverb level */
#define SPEEX_PREPROCESS_SET_DEREVERB_LEVEL 10
/** Get preprocessor dereverb level */
#define SPEEX_PREPROCESS_GET_DEREVERB_LEVEL 11
/** Set preprocessor dereverb decay */
#define SPEEX_PREPROCESS_SET_DEREVERB_DECAY 12
/** Get preprocessor dereverb decay */
#define SPEEX_PREPROCESS_GET_DEREVERB_DECAY 13
/** Set probability required for the VAD to go from silence to voice */
#define SPEEX_PREPROCESS_SET_PROB_START 14
/** Get probability required for the VAD to go from silence to voice */
#define SPEEX_PREPROCESS_GET_PROB_START 15
/** Set probability required for the VAD to stay in the voice state (integer percent) */
#define SPEEX_PREPROCESS_SET_PROB_CONTINUE 16
/** Get probability required for the VAD to stay in the voice state (integer percent) */
#define SPEEX_PREPROCESS_GET_PROB_CONTINUE 17
/** Set maximum attenuation of the noise in dB (negative number) */
#define SPEEX_PREPROCESS_SET_NOISE_SUPPRESS 18
/** Get maximum attenuation of the noise in dB (negative number) */
#define SPEEX_PREPROCESS_GET_NOISE_SUPPRESS 19
/** Set maximum attenuation of the residual echo in dB (negative number) */
#define SPEEX_PREPROCESS_SET_ECHO_SUPPRESS 20
/** Get maximum attenuation of the residual echo in dB (negative number) */
#define SPEEX_PREPROCESS_GET_ECHO_SUPPRESS 21
/** Set maximum attenuation of the residual echo in dB when near end is active (negative number) */
#define SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE 22
/** Get maximum attenuation of the residual echo in dB when near end is active (negative number) */
#define SPEEX_PREPROCESS_GET_ECHO_SUPPRESS_ACTIVE 23
/** Set the corresponding echo canceller state so that residual echo suppression can be performed (NULL for no residual echo suppression) */
#define SPEEX_PREPROCESS_SET_ECHO_STATE 24
/** Get the corresponding echo canceller state */
#define SPEEX_PREPROCESS_GET_ECHO_STATE 25
/** Set maximal gain increase in dB/second (int32) */
#define SPEEX_PREPROCESS_SET_AGC_INCREMENT 26
/** Get maximal gain increase in dB/second (int32) */
#define SPEEX_PREPROCESS_GET_AGC_INCREMENT 27
/** Set maximal gain decrease in dB/second (int32) */
#define SPEEX_PREPROCESS_SET_AGC_DECREMENT 28
/** Get maximal gain decrease in dB/second (int32) */
#define SPEEX_PREPROCESS_GET_AGC_DECREMENT 29
/** Set maximal gain in dB (int32) */
#define SPEEX_PREPROCESS_SET_AGC_MAX_GAIN 30
/** Get maximal gain in dB (int32) */
#define SPEEX_PREPROCESS_GET_AGC_MAX_GAIN 31
/* Can't set loudness */
/** Get loudness */
#define SPEEX_PREPROCESS_GET_AGC_LOUDNESS 33
/* Can't set gain */
/** Get current gain (int32 percent) */
#define SPEEX_PREPROCESS_GET_AGC_GAIN 35
/* Can't set spectrum size */
/** Get spectrum size for power spectrum (int32) */
#define SPEEX_PREPROCESS_GET_PSD_SIZE 37
/* Can't set power spectrum */
/** Get power spectrum (int32[] of squared values) */
#define SPEEX_PREPROCESS_GET_PSD 39
/* Can't set noise size */
/** Get spectrum size for noise estimate (int32) */
#define SPEEX_PREPROCESS_GET_NOISE_PSD_SIZE 41
/* Can't set noise estimate */
/** Get noise estimate (int32[] of squared values) */
#define SPEEX_PREPROCESS_GET_NOISE_PSD 43
/* Can't set speech probability */
/** Get speech probability in last frame (int32). */
#define SPEEX_PREPROCESS_GET_PROB 45
/** Set preprocessor Automatic Gain Control level (int32) */
#define SPEEX_PREPROCESS_SET_AGC_TARGET 46
/** Get preprocessor Automatic Gain Control level (int32) */
#define SPEEX_PREPROCESS_GET_AGC_TARGET 47
#ifdef __cplusplus
}
#endif
/** @}*/
#endif

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/* Copyright (C) 2007 Jean-Marc Valin
File: speex_resampler.h
Resampling code
The design goals of this code are:
- Very fast algorithm
- Low memory requirement
- Good *perceptual* quality (and not best SNR)
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:
1. Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
3. The name of the author may not be used to endorse or promote products
derived from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef SPEEX_RESAMPLER_H
#define SPEEX_RESAMPLER_H
#ifdef OUTSIDE_SPEEX
/********* WARNING: MENTAL SANITY ENDS HERE *************/
/* If the resampler is defined outside of Speex, we change the symbol names so that
there won't be any clash if linking with Speex later on. */
/* #define RANDOM_PREFIX your software name here */
#ifndef RANDOM_PREFIX
#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes"
#endif
#define CAT_PREFIX2(a,b) a ## b
#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b)
#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init)
#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac)
#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy)
#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float)
#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int)
#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float)
#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int)
#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate)
#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate)
#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac)
#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio)
#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality)
#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality)
#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride)
#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride)
#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride)
#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride)
#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency)
#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency)
#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros)
#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem)
#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror)
#define spx_int16_t short
#define spx_int32_t int
#define spx_uint16_t unsigned short
#define spx_uint32_t unsigned int
#else /* OUTSIDE_SPEEX */
#include "speex/speex_types.h"
#endif /* OUTSIDE_SPEEX */
#ifdef __cplusplus
extern "C" {
#endif
#define SPEEX_RESAMPLER_QUALITY_MAX 10
#define SPEEX_RESAMPLER_QUALITY_MIN 0
#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4
#define SPEEX_RESAMPLER_QUALITY_VOIP 3
#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5
enum {
RESAMPLER_ERR_SUCCESS = 0,
RESAMPLER_ERR_ALLOC_FAILED = 1,
RESAMPLER_ERR_BAD_STATE = 2,
RESAMPLER_ERR_INVALID_ARG = 3,
RESAMPLER_ERR_PTR_OVERLAP = 4,
RESAMPLER_ERR_MAX_ERROR
};
struct SpeexResamplerState_;
typedef struct SpeexResamplerState_ SpeexResamplerState;
/** Create a new resampler with integer input and output rates.
* @param nb_channels Number of channels to be processed
* @param in_rate Input sampling rate (integer number of Hz).
* @param out_rate Output sampling rate (integer number of Hz).
* @param quality Resampling quality between 0 and 10, where 0 has poor quality
* and 10 has very high quality.
* @return Newly created resampler state
* @retval NULL Error: not enough memory
*/
SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels,
spx_uint32_t in_rate,
spx_uint32_t out_rate,
int quality,
int *err);
/** Create a new resampler with fractional input/output rates. The sampling
* rate ratio is an arbitrary rational number with both the numerator and
* denominator being 32-bit integers.
* @param nb_channels Number of channels to be processed
* @param ratio_num Numerator of the sampling rate ratio
* @param ratio_den Denominator of the sampling rate ratio
* @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
* @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
* @param quality Resampling quality between 0 and 10, where 0 has poor quality
* and 10 has very high quality.
* @return Newly created resampler state
* @retval NULL Error: not enough memory
*/
SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
spx_uint32_t ratio_num,
spx_uint32_t ratio_den,
spx_uint32_t in_rate,
spx_uint32_t out_rate,
int quality,
int *err);
/** Destroy a resampler state.
* @param st Resampler state
*/
void speex_resampler_destroy(SpeexResamplerState *st);
/** Resample a float array. The input and output buffers must *not* overlap.
* @param st Resampler state
* @param channel_index Index of the channel to process for the multi-channel
* base (0 otherwise)
* @param in Input buffer
* @param in_len Number of input samples in the input buffer. Returns the
* number of samples processed
* @param out Output buffer
* @param out_len Size of the output buffer. Returns the number of samples written
*/
int speex_resampler_process_float(SpeexResamplerState *st,
spx_uint32_t channel_index,
const float *in,
spx_uint32_t *in_len,
float *out,
spx_uint32_t *out_len);
/** Resample an int array. The input and output buffers must *not* overlap.
* @param st Resampler state
* @param channel_index Index of the channel to process for the multi-channel
* base (0 otherwise)
* @param in Input buffer
* @param in_len Number of input samples in the input buffer. Returns the number
* of samples processed
* @param out Output buffer
* @param out_len Size of the output buffer. Returns the number of samples written
*/
int speex_resampler_process_int(SpeexResamplerState *st,
spx_uint32_t channel_index,
const spx_int16_t *in,
spx_uint32_t *in_len,
spx_int16_t *out,
spx_uint32_t *out_len);
/** Resample an interleaved float array. The input and output buffers must *not* overlap.
* @param st Resampler state
* @param in Input buffer
* @param in_len Number of input samples in the input buffer. Returns the number
* of samples processed. This is all per-channel.
* @param out Output buffer
* @param out_len Size of the output buffer. Returns the number of samples written.
* This is all per-channel.
*/
int speex_resampler_process_interleaved_float(SpeexResamplerState *st,
const float *in,
spx_uint32_t *in_len,
float *out,
spx_uint32_t *out_len);
/** Resample an interleaved int array. The input and output buffers must *not* overlap.
* @param st Resampler state
* @param in Input buffer
* @param in_len Number of input samples in the input buffer. Returns the number
* of samples processed. This is all per-channel.
* @param out Output buffer
* @param out_len Size of the output buffer. Returns the number of samples written.
* This is all per-channel.
*/
int speex_resampler_process_interleaved_int(SpeexResamplerState *st,
const spx_int16_t *in,
spx_uint32_t *in_len,
spx_int16_t *out,
spx_uint32_t *out_len);
/** Set (change) the input/output sampling rates (integer value).
* @param st Resampler state
* @param in_rate Input sampling rate (integer number of Hz).
* @param out_rate Output sampling rate (integer number of Hz).
*/
int speex_resampler_set_rate(SpeexResamplerState *st,
spx_uint32_t in_rate,
spx_uint32_t out_rate);
/** Get the current input/output sampling rates (integer value).
* @param st Resampler state
* @param in_rate Input sampling rate (integer number of Hz) copied.
* @param out_rate Output sampling rate (integer number of Hz) copied.
*/
void speex_resampler_get_rate(SpeexResamplerState *st,
spx_uint32_t *in_rate,
spx_uint32_t *out_rate);
/** Set (change) the input/output sampling rates and resampling ratio
* (fractional values in Hz supported).
* @param st Resampler state
* @param ratio_num Numerator of the sampling rate ratio
* @param ratio_den Denominator of the sampling rate ratio
* @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
* @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
*/
int speex_resampler_set_rate_frac(SpeexResamplerState *st,
spx_uint32_t ratio_num,
spx_uint32_t ratio_den,
spx_uint32_t in_rate,
spx_uint32_t out_rate);
/** Get the current resampling ratio. This will be reduced to the least
* common denominator.
* @param st Resampler state
* @param ratio_num Numerator of the sampling rate ratio copied
* @param ratio_den Denominator of the sampling rate ratio copied
*/
void speex_resampler_get_ratio(SpeexResamplerState *st,
spx_uint32_t *ratio_num,
spx_uint32_t *ratio_den);
/** Set (change) the conversion quality.
* @param st Resampler state
* @param quality Resampling quality between 0 and 10, where 0 has poor
* quality and 10 has very high quality.
*/
int speex_resampler_set_quality(SpeexResamplerState *st,
int quality);
/** Get the conversion quality.
* @param st Resampler state
* @param quality Resampling quality between 0 and 10, where 0 has poor
* quality and 10 has very high quality.
*/
void speex_resampler_get_quality(SpeexResamplerState *st,
int *quality);
/** Set (change) the input stride.
* @param st Resampler state
* @param stride Input stride
*/
void speex_resampler_set_input_stride(SpeexResamplerState *st,
spx_uint32_t stride);
/** Get the input stride.
* @param st Resampler state
* @param stride Input stride copied
*/
void speex_resampler_get_input_stride(SpeexResamplerState *st,
spx_uint32_t *stride);
/** Set (change) the output stride.
* @param st Resampler state
* @param stride Output stride
*/
void speex_resampler_set_output_stride(SpeexResamplerState *st,
spx_uint32_t stride);
/** Get the output stride.
* @param st Resampler state copied
* @param stride Output stride
*/
void speex_resampler_get_output_stride(SpeexResamplerState *st,
spx_uint32_t *stride);
/** Get the latency in input samples introduced by the resampler.
* @param st Resampler state
*/
int speex_resampler_get_input_latency(SpeexResamplerState *st);
/** Get the latency in output samples introduced by the resampler.
* @param st Resampler state
*/
int speex_resampler_get_output_latency(SpeexResamplerState *st);
/** Make sure that the first samples to go out of the resamplers don't have
* leading zeros. This is only useful before starting to use a newly created
* resampler. It is recommended to use that when resampling an audio file, as
* it will generate a file with the same length. For real-time processing,
* it is probably easier not to use this call (so that the output duration
* is the same for the first frame).
* @param st Resampler state
*/
int speex_resampler_skip_zeros(SpeexResamplerState *st);
/** Reset a resampler so a new (unrelated) stream can be processed.
* @param st Resampler state
*/
int speex_resampler_reset_mem(SpeexResamplerState *st);
/** Returns the English meaning for an error code
* @param err Error code
* @return English string
*/
const char *speex_resampler_strerror(int err);
#ifdef __cplusplus
}
#endif
#endif

91
branches/2.0/doubango/thirdparties/iphone/include/speex/speex_stereo.h Normal file
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@ -0,0 +1,91 @@
/* Copyright (C) 2002 Jean-Marc Valin*/
/**
@file speex_stereo.h
@brief Describes the handling for intensity stereo
*/
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef STEREO_H
#define STEREO_H
/** @defgroup SpeexStereoState SpeexStereoState: Handling Speex stereo files
* This describes the Speex intensity stereo encoding/decoding
* @{
*/
#include "speex/speex_types.h"
#include "speex/speex_bits.h"
#ifdef __cplusplus
extern "C" {
#endif
/** If you access any of these fields directly, I'll personally come and bite you */
typedef struct SpeexStereoState {
float balance; /**< Left/right balance info */
float e_ratio; /**< Ratio of energies: E(left+right)/[E(left)+E(right)] */
float smooth_left; /**< Smoothed left channel gain */
float smooth_right; /**< Smoothed right channel gain */
float reserved1; /**< Reserved for future use */
float reserved2; /**< Reserved for future use */
} SpeexStereoState;
/** Deprecated. Use speex_stereo_state_init() instead. */
#define SPEEX_STEREO_STATE_INIT {1,.5,1,1,0,0}
/** Initialise/create a stereo stereo state */
SpeexStereoState *speex_stereo_state_init();
/** Reset/re-initialise an already allocated stereo state */
void speex_stereo_state_reset(SpeexStereoState *stereo);
/** Destroy a stereo stereo state */
void speex_stereo_state_destroy(SpeexStereoState *stereo);
/** Transforms a stereo frame into a mono frame and stores intensity stereo info in 'bits' */
void speex_encode_stereo(float *data, int frame_size, SpeexBits *bits);
/** Transforms a stereo frame into a mono frame and stores intensity stereo info in 'bits' */
void speex_encode_stereo_int(spx_int16_t *data, int frame_size, SpeexBits *bits);
/** Transforms a mono frame into a stereo frame using intensity stereo info */
void speex_decode_stereo(float *data, int frame_size, SpeexStereoState *stereo);
/** Transforms a mono frame into a stereo frame using intensity stereo info */
void speex_decode_stereo_int(spx_int16_t *data, int frame_size, SpeexStereoState *stereo);
/** Callback handler for intensity stereo info */
int speex_std_stereo_request_handler(SpeexBits *bits, void *state, void *data);
#ifdef __cplusplus
}
#endif
/** @} */
#endif

133
branches/2.0/doubango/thirdparties/iphone/include/speex/speex_types.h Normal file
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@ -0,0 +1,133 @@
/* speex_types.h taken from libogg */
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: #ifdef jail to whip a few platforms into the UNIX ideal.
last mod: $Id: os_types.h 7524 2004-08-11 04:20:36Z conrad $
********************************************************************/
/**
@file speex_types.h
@brief Speex types
*/
#ifndef _SPEEX_TYPES_H
#define _SPEEX_TYPES_H
#if defined(_WIN32)
# if defined(__CYGWIN__)
# include <_G_config.h>
typedef _G_int32_t spx_int32_t;
typedef _G_uint32_t spx_uint32_t;
typedef _G_int16_t spx_int16_t;
typedef _G_uint16_t spx_uint16_t;
# elif defined(__MINGW32__)
typedef short spx_int16_t;
typedef unsigned short spx_uint16_t;
typedef int spx_int32_t;
typedef unsigned int spx_uint32_t;
# elif defined(__MWERKS__)
typedef int spx_int32_t;
typedef unsigned int spx_uint32_t;
typedef short spx_int16_t;
typedef unsigned short spx_uint16_t;
# else
/* MSVC/Borland */
typedef __int32 spx_int32_t;
typedef unsigned __int32 spx_uint32_t;
typedef __int16 spx_int16_t;
typedef unsigned __int16 spx_uint16_t;
# endif
#elif defined(__MACOS__)
# include <sys/types.h>
typedef SInt16 spx_int16_t;
typedef UInt16 spx_uint16_t;
typedef SInt32 spx_int32_t;
typedef UInt32 spx_uint32_t;
#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
# include <sys/types.h>
typedef int16_t spx_int16_t;
typedef u_int16_t spx_uint16_t;
typedef int32_t spx_int32_t;
typedef u_int32_t spx_uint32_t;
#elif defined(__BEOS__)
/* Be */
# include <inttypes.h>
typedef int16_t spx_int16_t;
typedef u_int16_t spx_uint16_t;
typedef int32_t spx_int32_t;
typedef u_int32_t spx_uint32_t;
#elif defined (__EMX__)
/* OS/2 GCC */
typedef short spx_int16_t;
typedef unsigned short spx_uint16_t;
typedef int spx_int32_t;
typedef unsigned int spx_uint32_t;
#elif defined (DJGPP)
/* DJGPP */
typedef short spx_int16_t;
typedef int spx_int32_t;
typedef unsigned int spx_uint32_t;
#elif defined(R5900)
/* PS2 EE */
typedef int spx_int32_t;
typedef unsigned spx_uint32_t;
typedef short spx_int16_t;
#elif defined(__SYMBIAN32__)
/* Symbian GCC */
typedef signed short spx_int16_t;
typedef unsigned short spx_uint16_t;
typedef signed int spx_int32_t;
typedef unsigned int spx_uint32_t;
#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
typedef short spx_int16_t;
typedef unsigned short spx_uint16_t;
typedef long spx_int32_t;
typedef unsigned long spx_uint32_t;
#elif defined(CONFIG_TI_C6X)
typedef short spx_int16_t;
typedef unsigned short spx_uint16_t;
typedef int spx_int32_t;
typedef unsigned int spx_uint32_t;
#elif ANDROID
typedef short spx_int16_t;
typedef unsigned short spx_uint16_t;
typedef int spx_int32_t;
typedef unsigned int spx_uint32_t;
#else
# include <speex/speex_config_types.h>
#endif
#endif /* _SPEEX_TYPES_H */

5
branches/2.0/doubango/tinyDAV/include/tinydav/audio/coreaudio/tdav_audiounit.h
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@ -33,11 +33,12 @@ TDAV_BEGIN_DECLS
typedef void* tdav_audiounit_handle_t;
tdav_audiounit_handle_t* tdav_audiounit_handle_create(uint64_t session_id);
AudioComponentInstance tdav_audiounit_handle_get_instance(tdav_audiounit_handle_t* self);
tdav_audiounit_handle_t* tdav_audiounit_handle_create(uint64_t session_id, uint32_t ptime);
/*inline*/ AudioComponentInstance tdav_audiounit_handle_get_instance(tdav_audiounit_handle_t* self);
int tdav_audiounit_handle_signal_consumer_prepared(tdav_audiounit_handle_t* self);
int tdav_audiounit_handle_signal_producer_prepared(tdav_audiounit_handle_t* self);
int tdav_audiounit_handle_start(tdav_audiounit_handle_t* self);
uint32_t tdav_audiounit_handle_get_frame_duration(tdav_audiounit_handle_t* self);
int tdav_audiounit_handle_stop(tdav_audiounit_handle_t* self);
int tdav_audiounit_handle_destroy(tdav_audiounit_handle_t** self);

17
branches/2.0/doubango/tinyDAV/include/tinydav/audio/coreaudio/tdav_consumer_audiounit.h
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@ -27,9 +27,12 @@
#if HAVE_COREAUDIO_AUDIO_UNIT
#include <AudioToolbox/AudioToolbox.h>
#include <speex/speex_buffer.h>
#include "tinydav/audio/coreaudio/tdav_audiounit.h"
#include "tinydav/audio/tdav_consumer_audio.h"
#include "tsk_mutex.h"
TDAV_BEGIN_DECLS
typedef struct tdav_consumer_audiounit_s
@ -37,8 +40,18 @@ typedef struct tdav_consumer_audiounit_s
TDAV_DECLARE_CONSUMER_AUDIO;
tdav_audiounit_handle_t* audioUnitHandle;
tsk_bool_t started;
tsk_bool_t paused;
unsigned started:1;
unsigned paused:1;
struct {
struct {
void* buffer;
tsk_size_t size;
} chunck;
SpeexBuffer* buffer;
tsk_size_t size;
tsk_mutex_handle_t* mutex;
} ring;
}
tdav_consumer_audiounit_t;

14
branches/2.0/doubango/tinyDAV/include/tinydav/audio/coreaudio/tdav_producer_audiounit.h
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@ -27,6 +27,7 @@
#if HAVE_COREAUDIO_AUDIO_UNIT
#include <AudioToolbox/AudioToolbox.h>
#include <speex/speex_buffer.h>
#include "tinydav/audio/coreaudio/tdav_audiounit.h"
#include "tinydav/audio/tdav_producer_audio.h"
@ -37,8 +38,17 @@ typedef struct tdav_producer_audiounit_s
TDAV_DECLARE_PRODUCER_AUDIO;
tdav_audiounit_handle_t* audioUnitHandle;
tsk_bool_t started;
tsk_bool_t paused;
unsigned started:1;
unsigned paused:1;
struct {
struct {
void* buffer;
tsk_size_t size;
} chunck;
SpeexBuffer* buffer;
tsk_size_t size;
} ring;
}
tdav_producer_audiounit_t;

2
branches/2.0/doubango/tinyDAV/include/tinydav/audio/tdav_speakup_jitterbuffer.h
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@ -44,7 +44,7 @@ typedef struct tdav_speakup_jitterBuffer_s
jitterbuffer *jbuffer;
uint8_t jcodec;
uint64_t ref_timestamp;
uint32_t framesize;
uint32_t frame_duration;
uint32_t rate;
}
tdav_speakup_jitterbuffer_t;

1
branches/2.0/doubango/tinyDAV/include/tinydav/audio/tdav_speex_jitterbuffer.h
View File

@ -45,6 +45,7 @@ typedef struct tdav_speex_jitterBuffer_s
JitterBuffer* state;
uint32_t rate;
uint32_t frame_duration;
}
tdav_speex_jitterbuffer_t;

66
branches/2.0/doubango/tinyDAV/include/tinydav/tdav_apple.h Normal file
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@ -0,0 +1,66 @@
/*
* Copyright (C) 2010-2011 Mamadou Diop.
*
* Contact: Mamadou Diop <diopmamadou(at)doubango.org>
*
* This file is part of Open Source Doubango Framework.
*
* DOUBANGO is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* DOUBANGO 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 DOUBANGO.
*
*/
#ifndef TDAV_APPLE_H
#define TDAV_APPLE_H
#if TDAV_UNDER_APPLE
#include <AudioToolbox/AudioToolbox.h>
#include "tsk_debug.h"
static int tdav_apple_init()
{
// initialize audio session
OSStatus status;
status = AudioSessionInitialize(NULL, NULL, NULL, NULL);
if(status){
TSK_DEBUG_ERROR("AudioSessionInitialize() failed with status code=%d", (int32_t)status);
return -1;
}
UInt32 audioCategory = kAudioSessionCategory_PlayAndRecord;
status = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(audioCategory), &audioCategory);
if(status){
TSK_DEBUG_ERROR("AudioSessionSetProperty(kAudioSessionProperty_AudioCategory) failed with status code=%d", (int32_t)status);
return -2;
}
// enable audio session
status = AudioSessionSetActive(true);
if(status){
TSK_DEBUG_ERROR("AudioSessionSetActive(true) failed with status code=%d", (int32_t)status);
return -3;
}
return 0;
}
static int tdav_apple_deinit()
{
// maybe other code use the session
// OSStatus status = AudioSessionSetActive(false);
return 0;
}
#endif /* TDAV_UNDER_APPLE */
#endif /* TDAV_APPLE_H */

2
branches/2.0/doubango/tinyDAV/include/tinydav/tdav_win32.h
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@ -37,7 +37,9 @@
TDAV_BEGIN_DECLS
int tdav_win32_init();
void tdav_win32_print_error(const char* func, HRESULT hr);
int tdav_win32_deinit();
TDAV_END_DECLS

11
branches/2.0/doubango/tinyDAV/include/tinydav_config.h
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@ -27,12 +27,16 @@
#undef _WIN32 /* Because of WINSCW */
#endif
/* Windows (XP/Vista/7/CE and Windows Mobile) macro definition.
*/
/* Windows (XP/Vista/7/CE and Windows Mobile) macro definition. */
#if defined(WIN32)|| defined(_WIN32) || defined(_WIN32_WCE)
# define TDAV_UNDER_WINDOWS 1
#endif
/* OS X or iOS */
#if defined(__APPLE__)
# define TDAV_UNDER_APPLE 1
#endif
#if (TDAV_UNDER_WINDOWS || defined(__SYMBIAN32__)) && defined(TINYDAV_EXPORTS)
# define TINYDAV_API __declspec(dllexport)
# define TINYDAV_GEXTERN __declspec(dllexport)
@ -44,8 +48,7 @@
# define TINYDAV_GEXTERN extern
#endif
/* Guards against C++ name mangling
*/
/* Guards against C++ name mangling */
#ifdef __cplusplus
# define TDAV_BEGIN_DECLS extern "C" {
# define TDAV_END_DECLS }

57
branches/2.0/doubango/tinyDAV/src/audio/coreaudio/tdav_audiounit.c
View File

@ -37,6 +37,7 @@ typedef struct tdav_audiounit_instance_s
{
TSK_DECLARE_OBJECT;
uint64_t session_id;
uint32_t frame_duration;
AudioComponentInstance audioUnit;
struct{
unsigned consumer:1;
@ -84,7 +85,7 @@ static int _tdav_audiounit_handle_signal_xxx_prepared(tdav_audiounit_handle_t* s
return 0;
}
tdav_audiounit_handle_t* tdav_audiounit_handle_create(uint64_t session_id)
tdav_audiounit_handle_t* tdav_audiounit_handle_create(uint64_t session_id, uint32_t ptime)
{
tdav_audiounit_instance_t* inst = tsk_null;
const tsk_list_item_t* item;
@ -96,11 +97,14 @@ tdav_audiounit_handle_t* tdav_audiounit_handle_create(uint64_t session_id)
audioDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
audioDescription.componentFlags = 0;
audioDescription.componentFlagsMask = 0;
__audioSystem = AudioComponentFindNext(NULL, &audioDescription);
}
if(!__audioSystem){
TSK_DEBUG_ERROR("Failed to find new audio component");
goto done;
if((__audioSystem = AudioComponentFindNext(NULL, &audioDescription))){
// leave blank
}
else {
TSK_DEBUG_ERROR("Failed to find new audio component");
goto done;
}
}
// create list used to hold instances
if(!__audioUnitInstances && !(__audioUnitInstances = tsk_list_create())){
@ -123,11 +127,44 @@ tdav_audiounit_handle_t* tdav_audiounit_handle_create(uint64_t session_id)
if((inst = tsk_object_new(tdav_audiounit_instance_def_t))){
OSStatus status;
tdav_audiounit_instance_t* _inst;
// set preferred buffer size
Float32 preferredBufferSize = (Float32)ptime / 1000.f; // in seconds
status = AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(preferredBufferSize), &preferredBufferSize);
if(status){
TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioOutputUnitProperty_SetInputCallback) failed with status=%d", (int32_t)status);
TSK_OBJECT_SAFE_FREE(inst);
goto done;
}
UInt32 size = sizeof(preferredBufferSize);
status = AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration, &size, &preferredBufferSize);
if(!status){
inst->frame_duration = (preferredBufferSize * 1000);
TSK_DEBUG_INFO("Frame duration=%d", inst->frame_duration);
}
else {
TSK_DEBUG_ERROR("AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration, %f) failed", preferredBufferSize);
}
// create new instance
if((status= AudioComponentInstanceNew(__audioSystem, &inst->audioUnit))){
TSK_DEBUG_ERROR("AudioComponentInstanceNew() failed with status=%d", (int32_t)status);
TSK_OBJECT_SAFE_FREE(inst);
goto done;
}
// enable all even if we know that it's already done by default
static UInt32 kOne = 1;
static UInt32 kZero = 0;
// static UInt32 kVoiceQuality = 127;
#define kInputBus 1
status = AudioUnitSetProperty(inst->audioUnit, kAUVoiceIOProperty_BypassVoiceProcessing,
kAudioUnitScope_Global, kInputBus, &kZero, sizeof(kZero));
status = AudioUnitSetProperty(inst->audioUnit, kAUVoiceIOProperty_VoiceProcessingEnableAGC,
kAudioUnitScope_Global, kInputBus, &kOne, sizeof(kOne));
status = AudioUnitSetProperty(inst->audioUnit, kAUVoiceIOProperty_DuckNonVoiceAudio,
kAudioUnitScope_Global, kInputBus, &kOne, sizeof(kOne));
// status = AudioUnitSetProperty(inst->audioUnit, kAUVoiceIOProperty_VoiceProcessingQuality,
// kAudioUnitScope_Global, kInputBus, &kVoiceQuality, sizeof(kVoiceQuality));
_inst = inst, _inst->session_id = session_id;
tsk_list_push_back_data(__audioUnitInstances, (void**)&_inst);
}
@ -175,6 +212,14 @@ int tdav_audiounit_handle_start(tdav_audiounit_handle_t* self)
return status ? -2 : 0;
}
uint32_t tdav_audiounit_handle_get_frame_duration(tdav_audiounit_handle_t* self)
{
if(self){
return ((tdav_audiounit_instance_t*)self)->frame_duration;
}
return 0;
}
int tdav_audiounit_handle_stop(tdav_audiounit_handle_t* self)
{
tdav_audiounit_instance_t* inst = (tdav_audiounit_instance_t*)self;

27
branches/2.0/doubango/tinyDAV/src/audio/coreaudio/tdav_consumer_audioqueue.c
View File

@ -40,26 +40,21 @@
#include "tsk_debug.h"
static void __handle_output_buffer(void *userdata, AudioQueueRef queue, AudioQueueBufferRef buffer) {
OSStatus ret;
void *data;
tsk_size_t out_size = 0;
tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)userdata;
if (!consumer->started) {
return;
}
if((data = tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(consumer), &out_size))){
// If we can get audio to play, then copy in the buffer
memcpy(buffer->mAudioData, data, TSK_MIN(consumer->buffer_size, out_size));
TSK_FREE(data);
} else{
// Put silence if there is no audio to play
memset(buffer->mAudioData, 0, consumer->buffer_size);
if(!tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(consumer), buffer->mAudioData, consumer->buffer_size)){
// Put silence
memset(buffer->mAudioData, 0, consumer->buffer_size);
}
// Re-enqueue the buffer
ret = AudioQueueEnqueueBuffer(consumer->queue, buffer, 0, NULL);
AudioQueueEnqueueBuffer(consumer->queue, buffer, 0, NULL);
// alert the jitter buffer
tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(consumer));
}
/* ============ Media Consumer Interface ================= */
@ -87,17 +82,17 @@ int tdav_consumer_audioqueue_prepare(tmedia_consumer_t* self, const tmedia_codec
// Create the audio stream description
AudioStreamBasicDescription *description = &(consumer->description);
description->mSampleRate = TDAV_CONSUMER_AUDIO(consumer)->output_rate ? TDAV_CONSUMER_AUDIO(consumer)->output_rate : TDAV_CONSUMER_AUDIO(consumer)->codec_rate;
description->mSampleRate = TMEDIA_CONSUMER(consumer)->audio.out.rate ? TMEDIA_CONSUMER(consumer)->audio.out.rate : TMEDIA_CONSUMER(consumer)->audio.in.rate;
description->mFormatID = kAudioFormatLinearPCM;
description->mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
description->mChannelsPerFrame = TDAV_CONSUMER_AUDIO(consumer)->channels;
description->mChannelsPerFrame = TMEDIA_CONSUMER(consumer)->audio.in.channels;
description->mFramesPerPacket = 1;
description->mBitsPerChannel = TDAV_CONSUMER_AUDIO(consumer)->bits_per_sample;
description->mBitsPerChannel = TMEDIA_CONSUMER(consumer)->audio.bits_per_sample;
description->mBytesPerPacket = description->mBitsPerChannel / 8 * description->mChannelsPerFrame;
description->mBytesPerFrame = description->mBytesPerPacket;
description->mReserved = 0;
int packetperbuffer = 1000 / TDAV_CONSUMER_AUDIO(consumer)->ptime;
int packetperbuffer = 1000 / TMEDIA_CONSUMER(consumer)->audio.ptime;
consumer->buffer_size = description->mSampleRate * description->mBytesPerFrame / packetperbuffer;
// Create the playback audio queue
@ -155,7 +150,7 @@ int tdav_consumer_audioqueue_consume(tmedia_consumer_t* self, const void* buffer
{
tdav_consumer_audioqueue_t* consumer = (tdav_consumer_audioqueue_t*)self;
if(!consumer || !buffer || !*buffer || !size){
if(!consumer || !buffer || !size){
TSK_DEBUG_ERROR("Invalid parameter");
return -1;
}

152
branches/2.0/doubango/tinyDAV/src/audio/coreaudio/tdav_consumer_audiounit.c
View File

@ -25,7 +25,22 @@
#if HAVE_COREAUDIO_AUDIO_UNIT
// AudioUnit already contains denoiser
#if HAVE_SPEEX_DSP && (!defined(HAVE_SPEEX_DENOISE) || HAVE_SPEEX_DENOISE)
# error "AudioUnit already contain denoiser => Disable it"
#endif
// FIXME: ONLY WORKS with SpeexJitterBuffer => add ifdef elif...
#undef DISABLE_JITTER_BUFFER
#define DISABLE_JITTER_BUFFER 0 // FIXME: Find why there is too delay when we use speex jitter buffer
#include "tsk_debug.h"
#include "tsk_memory.h"
#define kNoDataError -1
#define kRingPacketCount +5
static tsk_size_t tdav_consumer_audiounit_get(tdav_consumer_audiounit_t* self, void* data, tsk_size_t size);
static OSStatus __handle_output_buffer(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
@ -33,7 +48,50 @@ static OSStatus __handle_output_buffer(void *inRefCon,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
return noErr;
OSStatus status = noErr;
// tsk_size_t out_size;
tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t* )inRefCon;
if(!consumer->started || consumer->paused){
goto done;
}
if(!ioData){
TSK_DEBUG_ERROR("Invalid argument");
status = kNoDataError;
goto done;
}
// read from jitter buffer and fill ioData buffers
for(int i=0; i<ioData->mNumberBuffers; i++){
/* int ret = */ tdav_consumer_audiounit_get(consumer, ioData->mBuffers[i].mData, ioData->mBuffers[i].mDataByteSize);
}
done:
#if !DISABLE_JITTER_BUFFER
// alert the jitter buffer
tdav_consumer_audio_tick(TDAV_CONSUMER_AUDIO(consumer));
#endif
return status;
}
static tsk_size_t tdav_consumer_audiounit_get(tdav_consumer_audiounit_t* self, void* data, tsk_size_t size)
{
tsk_ssize_t retSize = 0;
#if DISABLE_JITTER_BUFFER
retSize = speex_buffer_read(self->ring.buffer, data, size);
if(retSize < size){
memset(((uint8_t*)data)+retSize, 0, (size - retSize));
}
#else
retSize = (tsk_ssize_t)tdav_consumer_audio_get(TDAV_CONSUMER_AUDIO(self), data, 370/*TSK_MIN(self->ring.chunck.size, size)*/);
printf("size=%d\n", size);
if(retSize < size){
memset(((uint8_t*)data)+retSize, 0, (size - retSize));
}
#endif
return retSize;
}
/* ============ Media Consumer Interface ================= */
@ -42,6 +100,7 @@ static OSStatus __handle_output_buffer(void *inRefCon,
static int tdav_consumer_audiounit_prepare(tmedia_consumer_t* self, const tmedia_codec_t* codec)
{
static UInt32 flagOne = 1;
AudioStreamBasicDescription audioFormat;
#define kOutputBus 0
tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self;
@ -55,7 +114,7 @@ static int tdav_consumer_audiounit_prepare(tmedia_consumer_t* self, const tmedia
TSK_DEBUG_ERROR("Already propared");
return -2;
}
if(!(consumer->audioUnitHandle = tdav_audiounit_handle_create(TMEDIA_CONSUMER(consumer)->session_id))){
if(!(consumer->audioUnitHandle = tdav_audiounit_handle_create(TMEDIA_CONSUMER(consumer)->session_id, codec->plugin->audio.ptime))){
TSK_DEBUG_ERROR("Failed to get audio unit instance for session with id=%lld", TMEDIA_CONSUMER(consumer)->session_id);
return -3;
}
@ -72,12 +131,12 @@ static int tdav_consumer_audiounit_prepare(tmedia_consumer_t* self, const tmedia
return -4;
}
else {
TMEDIA_CONSUMER(consumer)->audio.ptime = codec->plugin->audio.ptime;
uint32_t frame_duration = tdav_audiounit_handle_get_frame_duration(consumer->audioUnitHandle);
TMEDIA_CONSUMER(consumer)->audio.ptime = frame_duration ? frame_duration : codec->plugin->audio.ptime;
TMEDIA_CONSUMER(consumer)->audio.in.channels = codec->plugin->audio.channels;
TMEDIA_CONSUMER(consumer)->audio.in.rate = codec->plugin->rate;
// set format
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = TMEDIA_CONSUMER(consumer)->audio.out.rate ? TMEDIA_CONSUMER(consumer)->audio.out.rate : TMEDIA_CONSUMER(consumer)->audio.in.rate;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
@ -93,6 +152,7 @@ static int tdav_consumer_audiounit_prepare(tmedia_consumer_t* self, const tmedia
kOutputBus,
&audioFormat,
sizeof(audioFormat));
if(status){
TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioUnitProperty_StreamFormat) failed with status=%d", (int32_t)status);
return -5;
@ -115,6 +175,46 @@ static int tdav_consumer_audiounit_prepare(tmedia_consumer_t* self, const tmedia
}
}
// allocate the chunck buffer and create the ring
int packetperbuffer = (1000 / TMEDIA_CONSUMER(consumer)->audio.ptime);
consumer->ring.chunck.size = audioFormat.mSampleRate * audioFormat.mBytesPerFrame / packetperbuffer;
consumer->ring.size = kRingPacketCount * consumer->ring.chunck.size;
if(!(consumer->ring.chunck.buffer = tsk_realloc(consumer->ring.chunck.buffer, consumer->ring.chunck.size))){
TSK_DEBUG_ERROR("Failed to allocate new buffer");
return -7;
}
if(!consumer->ring.buffer){
consumer->ring.buffer = speex_buffer_init(consumer->ring.size);
}
else {
int ret;
if((ret = speex_buffer_resize(consumer->ring.buffer, consumer->ring.size))){
TSK_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", consumer->ring.size, ret);
return ret;
}
}
if(!consumer->ring.buffer){
TSK_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", consumer->ring.size);
return -8;
}
if(!(consumer->ring.mutex = tsk_mutex_create())){
TSK_DEBUG_ERROR("Failed to create mutex");
return -9;
}
// set maximum frames per slice as buffer size
//UInt32 numFrames = (UInt32)consumer->ring.chunck.size;
//status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(consumer->audioUnitHandle),
// kAudioUnitProperty_MaximumFramesPerSlice,
// kAudioUnitScope_Global,
// 0,
// &numFrames,
// sizeof(numFrames));
//if(status){
// TSK_DEBUG_ERROR("AudioUnitSetProperty(kAudioUnitProperty_MaximumFramesPerSlice, %u) failed with status=%d", (unsigned)numFrames, (int32_t)status);
// return -6;
//}
TSK_DEBUG_INFO("AudioUnit producer prepared");
return tdav_audiounit_handle_signal_consumer_prepared(consumer->audioUnitHandle);
}
@ -148,9 +248,36 @@ static int tdav_consumer_audiounit_start(tmedia_consumer_t* self)
}
static int tdav_consumer_audiounit_consume(tmedia_consumer_t* self, const void* buffer, tsk_size_t size, const tsk_object_t* proto_hdr)
{
TSK_DEBUG_ERROR("Not implemented");
return -1;
{
tdav_consumer_audiounit_t* consumer = (tdav_consumer_audiounit_t*)self;
if(!consumer || !buffer || !size){
TSK_DEBUG_ERROR("Invalid parameter");
return -1;
}
#if DISABLE_JITTER_BUFFER
{
if(consumer->ring.buffer){
tsk_mutex_lock(consumer->ring.mutex);
speex_buffer_write(consumer->ring.buffer, (void*)buffer, size);
tsk_mutex_unlock(consumer->ring.mutex);
return 0;
}
return -2;
}
#else
{
tsk_mutex_lock(consumer->ring.mutex);
speex_buffer_write(consumer->ring.buffer, (void*)buffer, size);
int avail = speex_buffer_get_available(consumer->ring.buffer);
while (avail >= consumer->ring.chunck.size) {
speex_buffer_read(consumer->ring.buffer, consumer->ring.chunck.buffer, consumer->ring.chunck.size);
tdav_consumer_audio_put(TDAV_CONSUMER_AUDIO(consumer), consumer->ring.chunck.buffer, consumer->ring.chunck.size, proto_hdr);
avail -= consumer->ring.chunck.size;
}
tsk_mutex_unlock(consumer->ring.mutex);
return 0;
}
#endif
}
static int tdav_consumer_audiounit_pause(tmedia_consumer_t* self)
@ -200,6 +327,7 @@ static tsk_object_t* tdav_consumer_audiounit_ctor(tsk_object_t * self, va_list *
if(consumer){
/* init base */
tdav_consumer_audio_init(TDAV_CONSUMER_AUDIO(consumer));
/* init self */
}
return self;
}
@ -208,6 +336,7 @@ static tsk_object_t* tdav_consumer_audiounit_dtor(tsk_object_t * self)
{
tdav_consumer_audiounit_t *consumer = self;
if(consumer){
/* deinit self */
// Stop the consumer if not done
if(consumer->started){
tdav_consumer_audiounit_stop(self);
@ -216,6 +345,13 @@ static tsk_object_t* tdav_consumer_audiounit_dtor(tsk_object_t * self)
if(consumer->audioUnitHandle){
tdav_audiounit_handle_destroy(&consumer->audioUnitHandle);
}
TSK_FREE(consumer->ring.chunck.buffer);
if(consumer->ring.buffer){
speex_buffer_destroy(consumer->ring.buffer);
}
if(consumer->ring.mutex){
tsk_mutex_destroy(&consumer->ring.mutex);
}
/* deinit base */
tdav_consumer_audio_deinit(TDAV_CONSUMER_AUDIO(consumer));

7
branches/2.0/doubango/tinyDAV/src/audio/coreaudio/tdav_producer_audioqueue.c
View File

@ -41,7 +41,6 @@
#include "tsk_debug.h"
static void __handle_input_buffer (void *userdata, AudioQueueRef queue, AudioQueueBufferRef buffer, const AudioTimeStamp *start_time, UInt32 number_packet_descriptions, const AudioStreamPacketDescription *packet_descriptions ) {
OSStatus ret;
tdav_producer_audioqueue_t* producer = (tdav_producer_audioqueue_t*)userdata;
if (!producer->started) {
@ -54,7 +53,7 @@ static void __handle_input_buffer (void *userdata, AudioQueueRef queue, AudioQue
}
// Re-enqueue the buffer
ret = AudioQueueEnqueueBuffer(producer->queue, buffer, 0, NULL);
AudioQueueEnqueueBuffer(producer->queue, buffer, 0, NULL);
}
/* ============ Media Producer Interface ================= */
@ -83,7 +82,7 @@ static int tdav_producer_audioqueue_prepare(tmedia_producer_t* self, const tmedi
// Create the audio stream description
AudioStreamBasicDescription *description = &(producer->description);
description->mSampleRate = TMEDIA_PRODUCER(producer)->audio.rate
description->mSampleRate = TMEDIA_PRODUCER(producer)->audio.rate;
description->mFormatID = kAudioFormatLinearPCM;
description->mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
description->mChannelsPerFrame = TMEDIA_PRODUCER(producer)->audio.channels;
@ -93,7 +92,7 @@ static int tdav_producer_audioqueue_prepare(tmedia_producer_t* self, const tmedi
description->mBytesPerFrame = description->mBytesPerPacket;
description->mReserved = 0;
int packetperbuffer = 1000 / TDAV_PRODUCER_AUDIO(producer)->ptime;
int packetperbuffer = 1000 / TMEDIA_PRODUCER(producer)->audio.ptime;
producer->buffer_size = description->mSampleRate * description->mBytesPerFrame / packetperbuffer;
// Create the record audio queue

93
branches/2.0/doubango/tinyDAV/src/audio/coreaudio/tdav_producer_audiounit.c
View File

@ -25,15 +25,57 @@
#if HAVE_COREAUDIO_AUDIO_UNIT
// AudioUnit already contains denoiser
#if HAVE_SPEEX_DSP && (!defined(HAVE_SPEEX_DENOISE) || HAVE_SPEEX_DENOISE)
# error "AudioUnit already contain denoiser => Disable it"
#endif
#include "tsk_memory.h"
#include "tsk_debug.h"
#define kRingPacketCount +3
static OSStatus __handle_input_buffer(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
return noErr;
OSStatus status = noErr;
tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)inRefCon;
// holder
AudioBuffer buffer;
buffer.mData = tsk_null;
buffer.mDataByteSize = 0;
// list of holders
AudioBufferList buffers;
buffers.mNumberBuffers = 1;
buffers.mBuffers[0] = buffer;
// render to get frames from the system
status = AudioUnitRender(tdav_audiounit_handle_get_instance(producer->audioUnitHandle),
ioActionFlags,
inTimeStamp,
inBusNumber,
inNumberFrames,
&buffers);
if(!status){
// Alert the session that there is new data to send
if(TMEDIA_PRODUCER(producer)->enc_cb.callback) {
speex_buffer_write(producer->ring.buffer, buffers.mBuffers[0].mData, buffers.mBuffers[0].mDataByteSize);
tsk_ssize_t avail = speex_buffer_get_available(producer->ring.buffer);
while (avail >= producer->ring.chunck.size) {
tsk_size_t read = speex_buffer_read(producer->ring.buffer, producer->ring.chunck.buffer, producer->ring.chunck.size);
TMEDIA_PRODUCER(producer)->enc_cb.callback(TMEDIA_PRODUCER(producer)->enc_cb.callback_data,
producer->ring.chunck.buffer, producer->ring.chunck.size);
avail -= read;
}
}
}
return status;
}
/* ============ Media Producer Interface ================= */
@ -42,11 +84,12 @@ static OSStatus __handle_input_buffer(void *inRefCon,
static int tdav_producer_audiounit_prepare(tmedia_producer_t* self, const tmedia_codec_t* codec)
{
static UInt32 flagOne = 1;
static UInt32 flagZero = 0;
// static UInt32 flagZero = 0;
#define kInputBus 1
tdav_producer_audiounit_t* producer = (tdav_producer_audiounit_t*)self;
OSStatus status;
AudioStreamBasicDescription audioFormat;
if(!producer || !codec || !codec->plugin){
TSK_DEBUG_ERROR("Invalid parameter");
@ -56,7 +99,7 @@ static int tdav_producer_audiounit_prepare(tmedia_producer_t* self, const tmedia
TSK_DEBUG_ERROR("Already propared");
return -2;
}
if(!(producer->audioUnitHandle = tdav_audiounit_handle_create(TMEDIA_PRODUCER(producer)->session_id))){
if(!(producer->audioUnitHandle = tdav_audiounit_handle_create(TMEDIA_PRODUCER(producer)->session_id, codec->plugin->audio.ptime))){
TSK_DEBUG_ERROR("Failed to get audio unit instance for session with id=%lld", TMEDIA_PRODUCER(producer)->session_id);
return -3;
}
@ -79,7 +122,6 @@ static int tdav_producer_audiounit_prepare(tmedia_producer_t* self, const tmedia
/* codec should have ptime */
// set format
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = TMEDIA_PRODUCER(producer)->audio.rate;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
@ -116,17 +158,41 @@ static int tdav_producer_audiounit_prepare(tmedia_producer_t* self, const tmedia
}
else {
// disbale buffer allocation as we will provide ours
status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle),
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flagZero,
sizeof(flagZero));
//status = AudioUnitSetProperty(tdav_audiounit_handle_get_instance(producer->audioUnitHandle),
// kAudioUnitProperty_ShouldAllocateBuffer,
// kAudioUnitScope_Output,
// kInputBus,
// &flagZero,
// sizeof(flagZero));
int packetperbuffer = (1000 / codec->plugin->audio.ptime);
producer->ring.chunck.size = audioFormat.mSampleRate * audioFormat.mBytesPerFrame / packetperbuffer;
// allocate our chunck buffer
if(!(producer->ring.chunck.buffer = tsk_realloc(producer->ring.chunck.buffer, producer->ring.chunck.size))){
TSK_DEBUG_ERROR("Failed to allocate new buffer");
return -7;
}
// create ringbuffer
producer->ring.size = kRingPacketCount * producer->ring.chunck.size;
if(!producer->ring.buffer){
producer->ring.buffer = speex_buffer_init(producer->ring.size);
}
else {
int ret;
if((ret = speex_buffer_resize(producer->ring.buffer, producer->ring.size))){
TSK_DEBUG_ERROR("speex_buffer_resize(%d) failed with error code=%d", producer->ring.size, ret);
return ret;
}
}
if(!producer->ring.buffer){
TSK_DEBUG_ERROR("Failed to create a new ring buffer with size = %d", producer->ring.size);
return -8;
}
}
}
}
TSK_DEBUG_INFO("AudioUnit producer prepared");
return tdav_audiounit_handle_signal_producer_prepared(producer->audioUnitHandle);;
}
@ -224,7 +290,10 @@ static tsk_object_t* tdav_producer_audiounit_dtor(tsk_object_t * self)
if (producer->audioUnitHandle) {
tdav_audiounit_handle_destroy(&producer->audioUnitHandle);
}
TSK_FREE(producer->ring.chunck.buffer);
if(producer->ring.buffer){
speex_buffer_destroy(producer->ring.buffer);
}
/* deinit base */
tdav_producer_audio_deinit(TDAV_PRODUCER_AUDIO(producer));
}

4
branches/2.0/doubango/tinyDAV/src/audio/tdav_consumer_audio.c
View File

@ -161,7 +161,9 @@ tsk_size_t tdav_consumer_audio_get(tdav_consumer_audio_t* self, void* out_data,
if(!TMEDIA_JITTER_BUFFER(self->jitterbuffer)->opened){
int ret;
if((ret = tmedia_jitterbuffer_open(TMEDIA_JITTER_BUFFER(self->jitterbuffer), TMEDIA_CONSUMER(self)->audio.ptime, TMEDIA_CONSUMER(self)->audio.in.rate))){
uint32_t frame_duration = TMEDIA_CONSUMER(self)->audio.ptime;
uint32_t rate = TMEDIA_CONSUMER(self)->audio.in.rate;
if((ret = tmedia_jitterbuffer_open(TMEDIA_JITTER_BUFFER(self->jitterbuffer), frame_duration, rate))){
TSK_DEBUG_ERROR("Failed to open jitterbuffer (%d)", ret);
tsk_safeobj_unlock(self);
return 0;

26
branches/2.0/doubango/tinyDAV/src/audio/tdav_session_audio.c
View File

@ -296,16 +296,14 @@ int tdav_session_audio_start(tmedia_session_t* self)
//trtp_manager_set_payload_type(audio->rtp_manager, codec->neg_format ? atoi(codec->neg_format) : atoi(codec->format));
ret = trtp_manager_start(audio->rtp_manager);
/* Consumer */
if(audio->consumer){
tmedia_consumer_prepare(audio->consumer, codec);
tmedia_consumer_start(audio->consumer);
}
/* Producer */
if(audio->producer){
tmedia_producer_prepare(audio->producer, codec);
tmedia_producer_start(audio->producer);
}
// because of AudioUnit under iOS => prepare both consumer and producer then start() at the same time
/* prepare consumer and producer */
if(audio->consumer) tmedia_consumer_prepare(audio->consumer, codec);
if(audio->producer) tmedia_producer_prepare(audio->producer, codec);
/* start consumer and producer */
if(audio->consumer) tmedia_consumer_start(audio->consumer);
if(audio->producer) tmedia_producer_start(audio->producer);
/* Denoise (AEC, Noise Suppression, AGC) */
if(audio->denoise && audio->encoder.codec){
tmedia_denoise_open(audio->denoise, TMEDIA_CODEC_PCM_FRAME_SIZE(audio->encoder.codec), TMEDIA_CODEC_RATE(audio->encoder.codec), tsk_true, 8000.0f, tsk_true, tsk_true);
@ -784,11 +782,15 @@ static tsk_object_t* tdav_session_audio_ctor(tsk_object_t * self, va_list * app)
if(session){
/* init base: called by tmedia_session_create() */
/* init self */
uint64_t session_id = TMEDIA_SESSION(session)->id;
tsk_safeobj_init(session);
if(!(session->consumer = tmedia_consumer_create(tdav_session_audio_plugin_def_t->type, TMEDIA_SESSION(session)->id))){
if(!session_id){ // set the session id if not already done
TMEDIA_SESSION(session)->id = session_id = tmedia_session_get_unique_id();
}
if(!(session->consumer = tmedia_consumer_create(tdav_session_audio_plugin_def_t->type, session_id))){
TSK_DEBUG_ERROR("Failed to create Audio consumer");
}
if((session->producer = tmedia_producer_create(tdav_session_audio_plugin_def_t->type, TMEDIA_SESSION(session)->id))){
if((session->producer = tmedia_producer_create(tdav_session_audio_plugin_def_t->type, session_id))){
tmedia_producer_set_enc_callback(session->producer, tdav_session_audio_producer_enc_cb, self);
}
else{

4
branches/2.0/doubango/tinyDAV/src/audio/tdav_speakup_jitterbuffer.c
View File

@ -49,7 +49,7 @@ static int _10ms_size_bytes = 160;
#define TDAV_SPEAKUP_10MS_FRAME_SIZE(self) (((self)->rate * TDAV_SPEAKUP_10MS)/1000)
#define TDAV_SPEAKUP_PTIME_FRAME_SIZE(self) (((self)->rate * (self)->framesize)/1000)
int tdav_speakup_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t framesize, uint32_t rate)
int tdav_speakup_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t frame_duration, uint32_t rate)
{
tdav_speakup_jitterbuffer_t *jitterbuffer = (tdav_speakup_jitterbuffer_t *)self;
if(!jitterbuffer->jbuffer){
@ -60,7 +60,7 @@ int tdav_speakup_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t framesi
jitterbuffer->jcodec = JB_CODEC_OTHER;
}
jitterbuffer->ref_timestamp = 0;
jitterbuffer->framesize = framesize;
jitterbuffer->frame_duration = frame_duration;
jitterbuffer->rate = rate;
return 0;

2
branches/2.0/doubango/tinyDAV/src/audio/tdav_speex_denoise.c
View File

@ -34,6 +34,8 @@
#include "tsk_memory.h"
#include "tsk_debug.h"
#include <string.h>
#define ECHO_TAIL 20
int tdav_speex_denoise_open(tmedia_denoise_t* self, uint32_t frame_size, uint32_t sampling_rate, tsk_bool_t denoise, float agc_level, tsk_bool_t aec, tsk_bool_t vad)

10
branches/2.0/doubango/tinyDAV/src/audio/tdav_speex_jitterbuffer.c
View File

@ -35,14 +35,15 @@
#include "tsk_debug.h"
int tdav_speex_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t framesize, uint32_t rate)
int tdav_speex_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t frame_duration, uint32_t rate)
{
tdav_speex_jitterbuffer_t *jitterbuffer = (tdav_speex_jitterbuffer_t *)self;
if(!(jitterbuffer->state = jitter_buffer_init((int)framesize))){
if(!(jitterbuffer->state = jitter_buffer_init((int)frame_duration))){
TSK_DEBUG_ERROR("jitter_buffer_init() failed");
return -2;
}
jitterbuffer->rate = rate;
jitterbuffer->frame_duration = frame_duration;
return 0;
}
@ -78,7 +79,7 @@ int tdav_speex_jitterbuffer_put(tmedia_jitterbuffer_t* self, void* data, tsk_siz
jb_packet.data = data;
jb_packet.len = data_size;
jb_packet.span = (data_size*500)/jitterbuffer->rate;
jb_packet.timestamp = (rtp_hdr->seq_num*jb_packet.span);
jb_packet.timestamp = (rtp_hdr->seq_num * jb_packet.span);
jb_packet.sequence = rtp_hdr->seq_num;
jb_packet.user_data = 0;
@ -107,13 +108,14 @@ tsk_size_t tdav_speex_jitterbuffer_get(tmedia_jitterbuffer_t* self, void* out_da
if ((ret = jitter_buffer_get(jitterbuffer->state, &jb_packet, (out_size*500)/jitterbuffer->rate, tsk_null)) != JITTER_BUFFER_OK) {
switch(ret){
case JITTER_BUFFER_MISSING: /*TSK_DEBUG_INFO("JITTER_BUFFER_MISSING - %d", ret);*/ break;
case JITTER_BUFFER_MISSING: /*TSK_DEBUG_INFO("JITTER_BUFFER_MISSING - %d", ret);*/ break;
case JITTER_BUFFER_INSERTION: /*TSK_DEBUG_INFO("JITTER_BUFFER_INSERTION - %d", ret);*/ break;
default: TSK_DEBUG_INFO("jitter_buffer_get() failed - %d", ret);
}
// jitter_buffer_update_delay(jitterbuffer->state, &jb_packet, NULL);
return 0;
}
jitter_buffer_update_delay(jitterbuffer->state, &jb_packet, NULL);
return out_size;
}

36
branches/2.0/doubango/tinyDAV/src/tdav.c
View File

@ -28,6 +28,12 @@
*/
#include "tinydav/tdav.h"
#if TDAV_UNDER_WINDOWS
# include "tinydav/tdav_win32.h"
#elif TDAV_UNDER_APPLE
# include "tinydav/tdav_apple.h"
#endif
// Media Contents, ...
#include "tinymedia.h"
@ -94,6 +100,19 @@
int tdav_init()
{
int ret = 0;
/* === OS specific === */
#if TDAV_UNDER_WINDOWS
if ((ret = tdav_win32_init())) {
return ret;
}
#elif TDAV_UNDER_APPLE
if ((ret = tdav_apple_init())) {
return ret;
}
#endif
/* === Initialize ffmpeg === */
#if HAVE_FFMPEG
avcodec_init();
@ -219,7 +238,7 @@ int tdav_init()
tmedia_jitterbuffer_plugin_register(tdav_speakup_jitterbuffer_plugin_def_t);
#endif
return 0;
return ret;
}
typedef struct tdav_codec_decl_s{
@ -370,6 +389,19 @@ tsk_bool_t tdav_codec_is_supported(tdav_codec_id_t codec)
int tdav_deinit()
{
int ret = 0;
/* === OS specific === */
#if TDAV_UNDER_WINDOWS
if ((ret = tdav_win32_deinit())) {
return ret;
}
#elif TDAV_UNDER_APPLE
if ((ret = tdav_apple_deinit())) {
return ret;
}
#endif
/* === UnRegister media contents === */
tmedia_content_plugin_unregister_all();
@ -470,5 +502,5 @@ int tdav_deinit()
tmedia_jitterbuffer_plugin_unregister(tdav_speakup_jitterbuffer_plugin_def_t);
#endif
return 0;
return ret;
}

9
branches/2.0/doubango/tinyDAV/src/tdav_win32.c
View File

@ -32,6 +32,11 @@
#include "tsk_debug.h"
int tdav_win32_init()
{
return 0;
}
void tdav_win32_print_error(const char* func, HRESULT hr)
{
CHAR* message = tsk_null;
@ -54,5 +59,9 @@ void tdav_win32_print_error(const char* func, HRESULT hr)
LocalFree(message);
}
int tdav_win32_deinit()
{
return 0;
}
#endif /* TDAV_UNDER_WINDOWS */

4
branches/2.0/doubango/tinyMEDIA/include/tinymedia/tmedia_jitterbuffer.h
View File

@ -59,7 +59,7 @@ typedef struct tmedia_jitterbuffer_plugin_def_s
//! full description (usefull for debugging)
const char* desc;
int (* open) (tmedia_jitterbuffer_t*, uint32_t framesize, uint32_t rate);
int (* open) (tmedia_jitterbuffer_t*, uint32_t frame_duration, uint32_t rate);
int (* tick) (tmedia_jitterbuffer_t*);
int (* put) (tmedia_jitterbuffer_t*, void* data, tsk_size_t data_size, const tsk_object_t* proto_hdr);
tsk_size_t (* get) (tmedia_jitterbuffer_t*, void* out_data, tsk_size_t out_size);
@ -69,7 +69,7 @@ typedef struct tmedia_jitterbuffer_plugin_def_s
tmedia_jitterbuffer_plugin_def_t;
TINYMEDIA_API int tmedia_jitterbuffer_init(tmedia_jitterbuffer_t* self);
TINYMEDIA_API int tmedia_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t framesize, uint32_t rate);
TINYMEDIA_API int tmedia_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t frame_duration, uint32_t rate);
TINYMEDIA_API int tmedia_jitterbuffer_tick(tmedia_jitterbuffer_t* self);
TINYMEDIA_API int tmedia_jitterbuffer_put(tmedia_jitterbuffer_t* self, void* data, tsk_size_t data_size, const tsk_object_t* proto_hdr);
TINYMEDIA_API tsk_size_t tmedia_jitterbuffer_get(tmedia_jitterbuffer_t* self, void* out_data, tsk_size_t out_size);

3
branches/2.0/doubango/tinyMEDIA/include/tinymedia/tmedia_session.h
View File

@ -57,7 +57,7 @@ typedef struct tmedia_session_s
{
TSK_DECLARE_OBJECT;
//! unique id
//! unique id. If you want to modifiy this field then you must use @ref tmedia_session_get_unique_id()
uint64_t id;
//! session type
tmedia_type_t type;
@ -113,6 +113,7 @@ typedef struct tmedia_session_plugin_def_s
}
tmedia_session_plugin_def_t;
TINYMEDIA_API uint64_t tmedia_session_get_unique_id();
TINYMEDIA_API int tmedia_session_init(tmedia_session_t* self, tmedia_type_t type);
TINYMEDIA_API int tmedia_session_cmp(const tsk_object_t* sess1, const tsk_object_t* sess2);
TINYMEDIA_API int tmedia_session_plugin_register(const tmedia_session_plugin_def_t* plugin);

4
branches/2.0/doubango/tinyMEDIA/src/tmedia_jitterbuffer.c
View File

@ -40,7 +40,7 @@ int tmedia_jitterbuffer_init(tmedia_jitterbuffer_t* self)
return 0;
}
int tmedia_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t framesize, uint32_t rate)
int tmedia_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t frame_duration, uint32_t rate)
{
int ret;
@ -53,7 +53,7 @@ int tmedia_jitterbuffer_open(tmedia_jitterbuffer_t* self, uint32_t framesize, ui
return 0;
}
if((ret = self->plugin->open(self, framesize, rate))){
if((ret = self->plugin->open(self, frame_duration, rate))){
TSK_DEBUG_ERROR("Failed to open [%s] jitterbufferr", self->plugin->desc);
return ret;
}

10
branches/2.0/doubango/tinyMEDIA/src/tmedia_session.c
View File

@ -79,6 +79,11 @@ int __pred_find_session_by_type(const tsk_list_item_t *item, const void *type)
return -1;
}
uint64_t tmedia_session_get_unique_id(){
static uint64_t __UniqueId = 1; // MUST not be equal to zero
return __UniqueId++;
}
/**@ingroup tmedia_session_group
* Initializes a newly created media session.
* @param self the media session to initialize.
@ -88,7 +93,6 @@ int __pred_find_session_by_type(const tsk_list_item_t *item, const void *type)
int tmedia_session_init(tmedia_session_t* self, tmedia_type_t type)
{
int ret = 0;
static uint64_t __UniqueId = 1;
if(!self){
TSK_DEBUG_ERROR("Invalid parameter");
@ -97,7 +101,9 @@ int tmedia_session_init(tmedia_session_t* self, tmedia_type_t type)
if(!self->initialized){
/* set values */
self->id = ++__UniqueId;
if(!self->id){
self->id = tmedia_session_get_unique_id();
}
self->type = type;
self->initialized = tsk_true;
self->bl = tmedia_bl_low;

2
branches/2.0/doubango/tinySIP/src/dialogs/tsip_dialog_invite.client.c
View File

@ -186,7 +186,7 @@ int c0000_Outgoing_2_Connected_X_i2xxINVITE(va_list *app)
tsip_dialog_invite_t *self = va_arg(*app, tsip_dialog_invite_t *);
const tsip_response_t *r2xxINVITE = va_arg(*app, const tsip_response_t *);
const tsip_action_t* action = va_arg(*app, const tsip_action_t *);
/* const tsip_action_t* action = */ va_arg(*app, const tsip_action_t *);
/* Update the dialog state */
if((ret = tsip_dialog_update(TSIP_DIALOG(self), r2xxINVITE))){