From 21410c238df235490b3958fec6ac51377dff3f3b Mon Sep 17 00:00:00 2001
From: Tristan Matthews <tristan.matthews@savoirfairelinux.com>
Date: Fri, 14 Sep 2012 16:41:20 -0400
Subject: [PATCH] * #15528: sip/audiortp: support asymmetric audio codecs
A call can send one codec and receive another.
---
.../audiortp/audio_rtp_record_handler.cpp | 81 ++++++++++++----
.../audio/audiortp/audio_rtp_record_handler.h | 4 +
.../src/audio/audiortp/audio_rtp_session.cpp | 3 +-
daemon/src/iax/iaxvoiplink.cpp | 2 +-
daemon/src/iax/iaxvoiplink.h | 2 +-
daemon/src/managerimpl.cpp | 2 +-
daemon/src/sip/sdp.cpp | 97 ++++++++++++-------
daemon/src/sip/sdp.h | 6 +-
daemon/src/sip/sipcall.cpp | 2 +-
daemon/src/sip/sipvoiplink.cpp | 44 +++++----
daemon/src/sip/sipvoiplink.h | 2 +-
daemon/src/voiplink.cpp | 1 +
daemon/src/voiplink.h | 2 +-
13 files changed, 168 insertions(+), 80 deletions(-)
diff --git a/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp b/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
index 4ab82a27c2..2735ccfb32 100644
--- a/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_record_handler.cpp
@@ -109,6 +109,7 @@ AudioRtpRecord::AudioRtpRecord() :
#endif
, dtmfPayloadType_(101) // same as Asterisk
, dead_(false)
+ , currentCodecIndex_(0)
{}
// Call from processData*
@@ -121,6 +122,16 @@ bool AudioRtpRecord::isDead()
#endif
}
+sfl::AudioCodec *
+AudioRtpRecord::getCurrentCodec() const
+{
+ if (audioCodecs_.empty() or currentCodecIndex_ >= audioCodecs_.size()) {
+ ERROR("No codec found");
+ return 0;
+ }
+ return audioCodecs_[currentCodecIndex_];
+}
+
void
AudioRtpRecord::deleteCodecs()
{
@@ -129,6 +140,23 @@ AudioRtpRecord::deleteCodecs()
audioCodecs_.clear();
}
+bool AudioRtpRecord::tryToSwitchPayloadTypes(int newPt)
+{
+ for (std::vector<AudioCodec *>::iterator i = audioCodecs_.begin(); i != audioCodecs_.end(); ++i)
+ if (*i and (*i)->getPayloadType() == newPt) {
+ codecPayloadType_ = (*i)->getPayloadType();
+ codecSampleRate_ = (*i)->getClockRate();
+ codecFrameSize_ = (*i)->getFrameSize();
+ hasDynamicPayloadType_ = (*i)->hasDynamicPayload();
+ currentCodecIndex_ = std::distance(audioCodecs_.begin(), i);
+ DEBUG("Switched payload type to %d", newPt);
+ return true;
+ }
+
+ ERROR("Could not switch payload types");
+ return false;
+}
+
AudioRtpRecord::~AudioRtpRecord()
{
dead_ = true;
@@ -174,6 +202,8 @@ void AudioRtpRecordHandler::setRtpMedia(const std::vector<AudioCodec*> &audioCod
audioRtpRecord_.deleteCodecs();
// Set varios codec info to reduce indirection
audioRtpRecord_.audioCodecs_ = audioCodecs;
+
+ audioRtpRecord_.currentCodecIndex_ = 0;
audioRtpRecord_.codecPayloadType_ = audioCodecs[0]->getPayloadType();
audioRtpRecord_.codecSampleRate_ = audioCodecs[0]->getClockRate();
audioRtpRecord_.codecFrameSize_ = audioCodecs[0]->getFrameSize();
@@ -273,13 +303,9 @@ int AudioRtpRecordHandler::processDataEncode()
{
ost::MutexLock lock(audioRtpRecord_.audioCodecMutex_);
- if (audioRtpRecord_.audioCodecs_.empty()) {
- ERROR("Audio codecs already destroyed");
- return 0;
- }
- RETURN_IF_NULL(audioRtpRecord_.audioCodecs_[0], 0, "Audio codec already destroyed");
+ RETURN_IF_NULL(audioRtpRecord_.getCurrentCodec(), 0, "Audio codec already destroyed");
unsigned char *micDataEncoded = audioRtpRecord_.encodedData_.data();
- return audioRtpRecord_.audioCodecs_[0]->encode(micDataEncoded, out, getCodecFrameSize());
+ return audioRtpRecord_.getCurrentCodec()->encode(micDataEncoded, out, getCodecFrameSize());
}
}
#undef RETURN_IF_NULL
@@ -291,27 +317,24 @@ void AudioRtpRecordHandler::processDataDecode(unsigned char *spkrData, size_t si
if (audioRtpRecord_.isDead())
return;
if (audioRtpRecord_.codecPayloadType_ != payloadType) {
- if (!warningInterval_) {
- warningInterval_ = 250;
- WARN("Invalid payload type %d, expected %d", payloadType, audioRtpRecord_.codecPayloadType_);
- WARN("We have %u codecs total", audioRtpRecord_.audioCodecs_.size());
+ const bool switched = audioRtpRecord_.tryToSwitchPayloadTypes(payloadType);
+ if (not switched) {
+ if (!warningInterval_) {
+ warningInterval_ = 250;
+ WARN("Invalid payload type %d, expected %d", payloadType, audioRtpRecord_.codecPayloadType_);
+ }
+ warningInterval_--;
+ return;
}
- warningInterval_--;
- return;
}
int inSamples = 0;
size = std::min(size, audioRtpRecord_.decData_.size());
SFLDataFormat *spkrDataDecoded = audioRtpRecord_.decData_.data();
{
- ost::MutexLock lock(audioRtpRecord_.audioCodecMutex_);
- if (audioRtpRecord_.audioCodecs_.empty()) {
- ERROR("Audio codecs already destroyed");
- return;
- }
- RETURN_IF_NULL(audioRtpRecord_.audioCodecs_[0], "Audio codecs already destroyed");
+ RETURN_IF_NULL(audioRtpRecord_.getCurrentCodec(), "Audio codecs already destroyed");
// Return the size of data in samples
- inSamples = audioRtpRecord_.audioCodecs_[0]->decode(spkrDataDecoded, spkrData, size);
+ inSamples = audioRtpRecord_.getCurrentCodec()->decode(spkrDataDecoded, spkrData, size);
}
#if HAVE_SPEEXDSP
@@ -354,6 +377,7 @@ void AudioRtpRecord::fadeInDecodedData(size_t size)
if (fadeFactor_ >= 1.0 or size > decData_.size())
return;
+ // FIXME: this takes a lot more cycles than a plain old loop
std::transform(decData_.begin(), decData_.begin() + size, decData_.begin(),
std::bind1st(std::multiplies<double>(), fadeFactor_));
@@ -361,4 +385,23 @@ void AudioRtpRecord::fadeInDecodedData(size_t size)
const double FADEIN_STEP_SIZE = 4.0;
fadeFactor_ *= FADEIN_STEP_SIZE;
}
+
+bool
+AudioRtpRecordHandler::codecsDiffer(const std::vector<AudioCodec*> &codecs) const
+{
+ const std::vector<AudioCodec*> ¤t = audioRtpRecord_.audioCodecs_;
+ if (codecs.size() != current.size())
+ return true;
+ for (std::vector<AudioCodec*>::const_iterator i = codecs.begin(); i != codecs.end(); ++i) {
+ if (*i) {
+ bool matched = false;
+ for (std::vector<AudioCodec*>::const_iterator j = current.begin(); !matched and j != current.end(); ++j)
+ matched = (*i)->getPayloadType() == (*j)->getPayloadType();
+ if (not matched)
+ return true;
+ }
+ }
+ return false;
+}
+
}
diff --git a/daemon/src/audio/audiortp/audio_rtp_record_handler.h b/daemon/src/audio/audiortp/audio_rtp_record_handler.h
index 2f35664bb2..98695bf922 100644
--- a/daemon/src/audio/audiortp/audio_rtp_record_handler.h
+++ b/daemon/src/audio/audiortp/audio_rtp_record_handler.h
@@ -73,6 +73,8 @@ class AudioRtpRecord {
AudioRtpRecord();
~AudioRtpRecord();
void deleteCodecs();
+ bool tryToSwitchPayloadTypes(int newPt);
+ sfl::AudioCodec* getCurrentCodec() const;
std::string callId_;
int codecSampleRate_;
std::list<DTMFEvent> dtmfQueue_;
@@ -112,6 +114,7 @@ class AudioRtpRecord {
#else
ucommon::atomic::counter dead_;
#endif
+ size_t currentCodecIndex_;
};
@@ -180,6 +183,7 @@ class AudioRtpRecordHandler {
void putDtmfEvent(char digit);
protected:
+ bool codecsDiffer(const std::vector<AudioCodec*> &codecs) const;
AudioRtpRecord audioRtpRecord_;
private:
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.cpp b/daemon/src/audio/audiortp/audio_rtp_session.cpp
index a33811f491..e7eb4fbbae 100644
--- a/daemon/src/audio/audiortp/audio_rtp_session.cpp
+++ b/daemon/src/audio/audiortp/audio_rtp_session.cpp
@@ -65,7 +65,8 @@ void AudioRtpSession::updateSessionMedia(const std::vector<AudioCodec*> &audioCo
{
int lastSamplingRate = audioRtpRecord_.codecSampleRate_;
- setSessionMedia(audioCodecs);
+ if (codecsDiffer(audioCodecs))
+ setSessionMedia(audioCodecs);
Manager::instance().audioSamplingRateChanged(audioRtpRecord_.codecSampleRate_);
diff --git a/daemon/src/iax/iaxvoiplink.cpp b/daemon/src/iax/iaxvoiplink.cpp
index ac846e0980..aaf1333cf2 100644
--- a/daemon/src/iax/iaxvoiplink.cpp
+++ b/daemon/src/iax/iaxvoiplink.cpp
@@ -475,7 +475,7 @@ IAXVoIPLink::getCurrentVideoCodecName(Call * /*call*/) const
}
std::string
-IAXVoIPLink::getCurrentAudioCodecName(Call *c) const
+IAXVoIPLink::getCurrentAudioCodecNames(Call *c) const
{
IAXCall *call = static_cast<IAXCall*>(c);
sfl::Codec *audioCodec = Manager::instance().audioCodecFactory.getCodec(call->getAudioCodec());
diff --git a/daemon/src/iax/iaxvoiplink.h b/daemon/src/iax/iaxvoiplink.h
index 3c7bf7efb1..d7b53d9f9c 100644
--- a/daemon/src/iax/iaxvoiplink.h
+++ b/daemon/src/iax/iaxvoiplink.h
@@ -193,7 +193,7 @@ class IAXVoIPLink : public VoIPLink {
* @param id The call identifier
*/
virtual std::string getCurrentVideoCodecName(Call *c) const;
- virtual std::string getCurrentAudioCodecName(Call *c) const;
+ virtual std::string getCurrentAudioCodecNames(Call *c) const;
private:
NON_COPYABLE(IAXVoIPLink);
diff --git a/daemon/src/managerimpl.cpp b/daemon/src/managerimpl.cpp
index 8917a13c68..94a450581e 100644
--- a/daemon/src/managerimpl.cpp
+++ b/daemon/src/managerimpl.cpp
@@ -1874,7 +1874,7 @@ std::string ManagerImpl::getCurrentAudioCodecName(const std::string& id)
Call::CallState state = call->getState();
if (state == Call::ACTIVE or state == Call::CONFERENCING)
- codecName = link->getCurrentAudioCodecName(call);
+ codecName = link->getCurrentAudioCodecNames(call);
}
return codecName;
diff --git a/daemon/src/sip/sdp.cpp b/daemon/src/sip/sdp.cpp
index e868ea4dc6..737e93ce9a 100644
--- a/daemon/src/sip/sdp.cpp
+++ b/daemon/src/sip/sdp.cpp
@@ -71,15 +71,23 @@ Sdp::Sdp(pj_pool_t *pool)
, telephoneEventPayload_(101) // same as asterisk
{}
+namespace {
+ bool hasPayload(const std::vector<sfl::AudioCodec*> &codecs, int pt)
+ {
+ for (std::vector<sfl::AudioCodec*>::const_iterator i = codecs.begin(); i != codecs.end(); ++i)
+ if (*i and (*i)->getPayloadType() == pt)
+ return true;
+ return false;
+ }
+}
+
+
void Sdp::setActiveLocalSdpSession(const pjmedia_sdp_session *sdp)
{
activeLocalSession_ = (pjmedia_sdp_session*) sdp;
- if (activeLocalSession_->media_count < 1)
- return;
-
- for (unsigned media = 0; media < activeLocalSession_->media_count; ++media) {
- pjmedia_sdp_media *current = activeLocalSession_->media[media];
+ for (unsigned i = 0; i < activeLocalSession_->media_count; ++i) {
+ pjmedia_sdp_media *current = activeLocalSession_->media[i];
for (unsigned fmt = 0; fmt < current->desc.fmt_count; ++fmt) {
static const pj_str_t STR_RTPMAP = { (char*) "rtpmap", 6 };
@@ -93,24 +101,22 @@ void Sdp::setActiveLocalSdpSession(const pjmedia_sdp_session *sdp)
pjmedia_sdp_rtpmap *rtpmap;
pjmedia_sdp_attr_to_rtpmap(memPool_, attribute, &rtpmap);
- string type(current->desc.media.ptr, current->desc.media.slen);
- if (type == "audio") {
+ if (!pj_stricmp2(¤t->desc.media, "audio")) {
const int pt = pj_strtoul(&rtpmap->pt);
- sfl::Codec *codec = Manager::instance().audioCodecFactory.getCodec(pt);
- if (codec)
- sessionAudioMedia_.push_back(codec);
- else {
- DEBUG("Could not get codec for payload type %lu", pt);
- break;
+ if (not hasPayload(sessionAudioMedia_, pt)) {
+ sfl::AudioCodec *codec = Manager::instance().audioCodecFactory.getCodec(pt);
+ if (codec)
+ sessionAudioMedia_.push_back(codec);
+ else
+ ERROR("Could not get codec for payload type %lu", pt);
}
- } else if (type == "video")
+ } else if (!pj_stricmp2(¤t->desc.media, "video"))
sessionVideoMedia_.push_back(string(rtpmap->enc_name.ptr, rtpmap->enc_name.slen));
}
}
}
-
void Sdp::setActiveRemoteSdpSession(const pjmedia_sdp_session *sdp)
{
activeRemoteSession_ = (pjmedia_sdp_session*) sdp;
@@ -121,22 +127,44 @@ void Sdp::setActiveRemoteSdpSession(const pjmedia_sdp_session *sdp)
}
for (unsigned i = 0; i < sdp->media_count; i++) {
- if (pj_stricmp2(&sdp->media[i]->desc.media, "audio") == 0) {
- pjmedia_sdp_media *r_media = sdp->media[i];
+ pjmedia_sdp_media *r_media = sdp->media[i];
+ if (!pj_stricmp2(&r_media->desc.media, "audio")) {
static const pj_str_t STR_TELEPHONE_EVENT = { (char*) "telephone-event", 15};
- pjmedia_sdp_attr *attribute = pjmedia_sdp_attr_find(r_media->attr_count, r_media->attr, &STR_TELEPHONE_EVENT, NULL);
+ pjmedia_sdp_attr *telephoneEvent = pjmedia_sdp_attr_find(r_media->attr_count, r_media->attr, &STR_TELEPHONE_EVENT, NULL);
- if (attribute != NULL) {
+ if (telephoneEvent != NULL) {
pjmedia_sdp_rtpmap *rtpmap;
- pjmedia_sdp_attr_to_rtpmap(memPool_, attribute, &rtpmap);
+ pjmedia_sdp_attr_to_rtpmap(memPool_, telephoneEvent, &rtpmap);
telephoneEventPayload_ = pj_strtoul(&rtpmap->pt);
}
- return;
+ // add audio codecs from remote as needed
+ for (unsigned fmt = 0; fmt < r_media->desc.fmt_count; ++fmt) {
+
+ static const pj_str_t STR_RTPMAP = { (char*) "rtpmap", 6 };
+ pjmedia_sdp_attr *rtpMapAttr = pjmedia_sdp_media_find_attr(r_media, &STR_RTPMAP, NULL);
+
+ if (!rtpMapAttr) {
+ ERROR("Could not find rtpmap attribute");
+ continue;
+ }
+
+ pjmedia_sdp_rtpmap *rtpmap;
+ pjmedia_sdp_attr_to_rtpmap(memPool_, rtpMapAttr, &rtpmap);
+
+ const int pt = pj_strtoul(&rtpmap->pt);
+ if (not hasPayload(sessionAudioMedia_, pt)) {
+ sfl::AudioCodec *codec = Manager::instance().audioCodecFactory.getCodec(pt);
+ if (codec) {
+ DEBUG("Adding codec with new payload type %d", pt);
+ sessionAudioMedia_.push_back(codec);
+ } else
+ DEBUG("Could not get codec for payload type %lu", pt);
+ }
+ }
}
}
-
- ERROR("Could not found dtmf event from remote sdp");
+ DEBUG("Using %u audio codecs total", sessionAudioMedia_.size());
}
string Sdp::getSessionVideoCodec() const
@@ -148,20 +176,23 @@ string Sdp::getSessionVideoCodec() const
return sessionVideoMedia_[0];
}
-string Sdp::getAudioCodecName() const
+string Sdp::getAudioCodecNames() const
{
- sfl::AudioCodec *codec = getSessionAudioMedia();
- return codec ? codec->getMimeSubtype() : "";
+ std::string result;
+ char sep = ' ';
+ for (std::vector<sfl::AudioCodec*>::const_iterator i = sessionAudioMedia_.begin();
+ i != sessionAudioMedia_.end(); ++i) {
+ if (i == sessionAudioMedia_.end() - 1)
+ sep = '\0';
+ if (*i)
+ result += (*i)->getMimeSubtype() + sep;
+ }
+ return result;
}
-sfl::AudioCodec* Sdp::getSessionAudioMedia() const
+void Sdp::getSessionAudioMedia(std::vector<sfl::AudioCodec*> &codecs) const
{
- if (sessionAudioMedia_.empty()) {
- ERROR("No codec description for this media");
- return 0;
- }
-
- return static_cast<sfl::AudioCodec *>(sessionAudioMedia_[0]);
+ codecs = sessionAudioMedia_;
}
diff --git a/daemon/src/sip/sdp.h b/daemon/src/sip/sdp.h
index 90eef19a4b..2fefc3f876 100644
--- a/daemon/src/sip/sdp.h
+++ b/daemon/src/sip/sdp.h
@@ -239,9 +239,9 @@ class Sdp {
void setMediaTransportInfoFromRemoteSdp();
- std::string getAudioCodecName() const;
+ std::string getAudioCodecNames() const;
std::string getSessionVideoCodec() const;
- sfl::AudioCodec* getSessionAudioMedia() const;
+ void getSessionAudioMedia(std::vector<sfl::AudioCodec*> &) const;
// Sets @param settings with appropriate values and returns true if
// we are sending video, false otherwise
bool getOutgoingVideoSettings(std::map<std::string, std::string> &settings) const;
@@ -298,7 +298,7 @@ class Sdp {
/**
* The codecs that will be used by the session (after the SDP negotiation)
*/
- std::vector<sfl::Codec *> sessionAudioMedia_;
+ std::vector<sfl::AudioCodec *> sessionAudioMedia_;
std::vector<std::string> sessionVideoMedia_;
std::string localIpAddr_;
diff --git a/daemon/src/sip/sipcall.cpp b/daemon/src/sip/sipcall.cpp
index 1987fe677d..f91a00af42 100644
--- a/daemon/src/sip/sipcall.cpp
+++ b/daemon/src/sip/sipcall.cpp
@@ -89,7 +89,7 @@ std::map<std::string, std::string>
SIPCall::createHistoryEntry() const
{
std::map<std::string, std::string> entry(Call::createHistoryEntry());
- entry[HistoryItem::AUDIO_CODEC_KEY] = local_sdp_->getAudioCodecName();
+ entry[HistoryItem::AUDIO_CODEC_KEY] = local_sdp_->getAudioCodecNames();
#ifdef SFL_VIDEO
entry[HistoryItem::VIDEO_CODEC_KEY] = local_sdp_->getSessionVideoCodec();
#endif
diff --git a/daemon/src/sip/sipvoiplink.cpp b/daemon/src/sip/sipvoiplink.cpp
index 72f9fc4aa0..2c1182b213 100644
--- a/daemon/src/sip/sipvoiplink.cpp
+++ b/daemon/src/sip/sipvoiplink.cpp
@@ -1030,19 +1030,23 @@ SIPVoIPLink::offhold(const std::string& id)
throw VoipLinkException("Could not find sdp session");
try {
- int pl = PAYLOAD_CODEC_ULAW;
- sfl::AudioCodec *sessionMedia = sdpSession->getSessionAudioMedia();
- if (sessionMedia)
- pl = sessionMedia->getPayloadType();
+ std::vector<sfl::AudioCodec*> sessionMedia;
+ sdpSession->getSessionAudioMedia(sessionMedia);
+ std::vector<sfl::AudioCodec*> audioCodecs;
+ for (std::vector<sfl::AudioCodec*>::const_iterator i = sessionMedia.begin();
+ i != sessionMedia.end(); ++i) {
- // Create a new instance for this codec
- sfl::AudioCodec* ac = Manager::instance().audioCodecFactory.instantiateCodec(pl);
+ if (!*i)
+ continue;
- if (ac == NULL)
- throw VoipLinkException("Could not instantiate codec");
+ // Create a new instance for this codec
+ sfl::AudioCodec* ac = Manager::instance().audioCodecFactory.instantiateCodec((*i)->getPayloadType());
- std::vector<sfl::AudioCodec *> audioCodecs;
- audioCodecs.push_back(ac);
+ if (ac == NULL)
+ throw VoipLinkException("Could not instantiate codec");
+
+ audioCodecs.push_back(ac);
+ }
call->getAudioRtp().initConfig();
call->getAudioRtp().initSession();
@@ -1274,9 +1278,9 @@ SIPVoIPLink::getCurrentVideoCodecName(Call *call) const
}
std::string
-SIPVoIPLink::getCurrentAudioCodecName(Call *call) const
+SIPVoIPLink::getCurrentAudioCodecNames(Call *call) const
{
- return static_cast<SIPCall*>(call)->getLocalSDP()->getAudioCodecName();
+ return static_cast<SIPCall*>(call)->getLocalSDP()->getAudioCodecNames();
}
/* Only use this macro with string literals or character arrays, will not work
@@ -1751,8 +1755,9 @@ void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
}
#endif // HAVE_SDES
- sfl::AudioCodec *sessionMedia = sdpSession->getSessionAudioMedia();
- if (!sessionMedia)
+ std::vector<sfl::AudioCodec*> sessionMedia;
+ sdpSession->getSessionAudioMedia(sessionMedia);
+ if (sessionMedia.empty())
return;
try {
@@ -1760,16 +1765,19 @@ void sdp_media_update_cb(pjsip_inv_session *inv, pj_status_t status)
Manager::instance().getAudioDriver()->startStream();
Manager::instance().audioLayerMutexUnlock();
- int pl = sessionMedia->getPayloadType();
+ std::vector<AudioCodec*> audioCodecs;
+ for (std::vector<sfl::AudioCodec*>::const_iterator i = sessionMedia.begin(); i != sessionMedia.end(); ++i) {
+ if (!*i)
+ continue;
+ const int pl = (*i)->getPayloadType();
- if (pl != call->getAudioRtp().getSessionMedia()) {
sfl::AudioCodec *ac = Manager::instance().audioCodecFactory.instantiateCodec(pl);
if (!ac)
throw std::runtime_error("Could not instantiate codec");
- std::vector<AudioCodec*> audioCodecs;
audioCodecs.push_back(ac);
- call->getAudioRtp().updateSessionMedia(audioCodecs);
}
+ if (not audioCodecs.empty())
+ call->getAudioRtp().updateSessionMedia(audioCodecs);
} catch (const SdpException &e) {
ERROR("%s", e.what());
} catch (const std::exception &rtpException) {
diff --git a/daemon/src/sip/sipvoiplink.h b/daemon/src/sip/sipvoiplink.h
index 5dfad36f4c..d525a1c04c 100644
--- a/daemon/src/sip/sipvoiplink.h
+++ b/daemon/src/sip/sipvoiplink.h
@@ -252,7 +252,7 @@ class SIPVoIPLink : public VoIPLink {
* @param c The call identifier
*/
std::string getCurrentVideoCodecName(Call *c) const;
- std::string getCurrentAudioCodecName(Call *c) const;
+ std::string getCurrentAudioCodecNames(Call *c) const;
/**
* Retrive useragent name from account
diff --git a/daemon/src/voiplink.cpp b/daemon/src/voiplink.cpp
index 75f468ed1d..d6afe4954b 100644
--- a/daemon/src/voiplink.cpp
+++ b/daemon/src/voiplink.cpp
@@ -32,6 +32,7 @@
*/
#include "voiplink.h"
+#include "account.h"
VoIPLink::VoIPLink() : handlingEvents_(false) {}
diff --git a/daemon/src/voiplink.h b/daemon/src/voiplink.h
index 4cc8590b45..841eb7679b 100644
--- a/daemon/src/voiplink.h
+++ b/daemon/src/voiplink.h
@@ -147,7 +147,7 @@ class VoIPLink {
* @param call The call
*/
virtual std::string getCurrentVideoCodecName(Call *call) const = 0;
- virtual std::string getCurrentAudioCodecName(Call *call) const = 0;
+ virtual std::string getCurrentAudioCodecNames(Call *call) const = 0;
/**
* Send a message to a call identified by its callid
--
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