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* Copyright 2010, Google Inc.
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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#ifndef MSILBC_LIBRARY
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#define MSILBC_LIBRARY "/usr/lib/mediastreamer/plugins/libmsilbc.so"
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#define PORT_UNUSED -1
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// LinphoneMediaEngine is a Linphone implementation of MediaEngine
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#include <mediastreamer2/mediastream.h>
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#include <mediastreamer2/mssndcard.h>
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#include <mediastreamer2/msfilter.h>
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#include "talk/session/phone/linphonemediaengine.h"
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#include "talk/base/buffer.h"
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#include "talk/base/event.h"
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#include "talk/base/logging.h"
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#include "talk/base/pathutils.h"
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#include "talk/base/stream.h"
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#include "talk/session/phone/rtpdump.h"
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///////////////////////////////////////////////////////////////////////////
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// Implementation of LinphoneMediaEngine.
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///////////////////////////////////////////////////////////////////////////
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LinphoneMediaEngine::LinphoneMediaEngine(const std::string& ringWav, const std::string& callWav) : ring_wav_(ringWav), call_wav_(callWav) {
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char * path = strdup(MSILBC_LIBRARY);
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char * dirc = dirname(path);
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ms_load_plugins(dirc);
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if (ms_filter_codec_supported("iLBC"))
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if (ms_filter_codec_supported("speex"))
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if (ms_filter_codec_supported("gsm"))
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voice_codecs_.push_back(AudioCodec(110, payload_type_speex_wb.mime_type, payload_type_speex_wb.clock_rate, 0, 1, 8));
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voice_codecs_.push_back(AudioCodec(111, payload_type_speex_nb.mime_type, payload_type_speex_nb.clock_rate, 0, 1, 7));
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voice_codecs_.push_back(AudioCodec(102, payload_type_ilbc.mime_type, payload_type_ilbc.clock_rate, 0, 1, 4));
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voice_codecs_.push_back(AudioCodec(3, payload_type_gsm.mime_type, payload_type_gsm.clock_rate, 0, 1, 3));
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voice_codecs_.push_back(AudioCodec(0, payload_type_pcmu8000.mime_type, payload_type_pcmu8000.clock_rate, 0, 1, 2));
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voice_codecs_.push_back(AudioCodec(101, payload_type_telephone_event.mime_type, payload_type_telephone_event.clock_rate, 0, 1, 1));
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void LinphoneMediaEngine::Terminate() {
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int LinphoneMediaEngine::GetCapabilities() {
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int capabilities = 0;
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capabilities |= AUDIO_SEND;
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capabilities |= AUDIO_RECV;
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VoiceMediaChannel* LinphoneMediaEngine::CreateChannel() {
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return new LinphoneVoiceChannel(this);
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VideoMediaChannel* LinphoneMediaEngine::CreateVideoChannel(VoiceMediaChannel* voice_ch) {
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bool LinphoneMediaEngine::FindAudioCodec(const AudioCodec &c) {
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if (c.name == payload_type_telephone_event.mime_type)
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if (have_speex && c.name == payload_type_speex_wb.mime_type && c.clockrate == payload_type_speex_wb.clock_rate)
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if (have_speex && c.name == payload_type_speex_nb.mime_type && c.clockrate == payload_type_speex_nb.clock_rate)
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if (have_ilbc && c.name == payload_type_ilbc.mime_type)
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if (have_gsm && c.name == payload_type_gsm.mime_type)
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///////////////////////////////////////////////////////////////////////////
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// Implementation of LinphoneVoiceChannel.
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///////////////////////////////////////////////////////////////////////////
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LinphoneVoiceChannel::LinphoneVoiceChannel(LinphoneMediaEngine*eng)
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talk_base::Thread *thread = talk_base::ThreadManager::Instance()->CurrentThread();
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talk_base::SocketServer *ss = thread->socketserver();
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socket_.reset(ss->CreateAsyncSocket(SOCK_DGRAM));
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socket_->Bind(talk_base::SocketAddress("localhost", 0)); /* 0 means that OS will choose some free port */
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port1 = socket_->GetLocalAddress().port(); /* and here we get port choosed by OS */
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socket_->SignalReadEvent.connect(this, &LinphoneVoiceChannel::OnIncomingData);
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LinphoneVoiceChannel::~LinphoneVoiceChannel()
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audio_stream_stop(audio_stream_);
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bool LinphoneVoiceChannel::SetPlayout(bool playout) {
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bool LinphoneVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
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std::vector<AudioCodec>::const_iterator i;
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ortp_set_log_level_mask(ORTP_MESSAGE|ORTP_WARNING|ORTP_ERROR|ORTP_FATAL);
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for (i = codecs.begin(); i < codecs.end(); i++) {
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if (!engine_->FindAudioCodec(*i))
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if (engine_->have_ilbc && i->name == payload_type_ilbc.mime_type) {
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rtp_profile_set_payload(&av_profile, i->id, &payload_type_ilbc);
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} else if (engine_->have_speex && i->name == payload_type_speex_wb.mime_type && i->clockrate == payload_type_speex_wb.clock_rate) {
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rtp_profile_set_payload(&av_profile, i->id, &payload_type_speex_wb);
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} else if (engine_->have_speex && i->name == payload_type_speex_nb.mime_type && i->clockrate == payload_type_speex_nb.clock_rate) {
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rtp_profile_set_payload(&av_profile, i->id, &payload_type_speex_nb);
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} else if (engine_->have_gsm && i->name == payload_type_gsm.mime_type) {
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rtp_profile_set_payload(&av_profile, i->id, &payload_type_gsm);
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} else if (i->name == payload_type_telephone_event.mime_type) {
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rtp_profile_set_payload(&av_profile, i->id, &payload_type_telephone_event);
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} else if (i->id == 0)
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rtp_profile_set_payload(&av_profile, 0, &payload_type_pcmu8000);
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LOG(LS_INFO) << "Using " << i->name << "/" << i->clockrate;
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audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, i->id, 250, 0); /* -1 means that function will choose some free port */
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port2 = rtp_session_get_local_port(audio_stream_->session);
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// We're being asked to set an empty list of codecs. This will only happen when
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// working with a buggy client; let's try PCMU.
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LOG(LS_WARNING) << "Received empty list of codces; using PCMU/8000";
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audio_stream_ = audio_stream_start(&av_profile, -1, "localhost", port1, 0, 250, 0); /* -1 means that function will choose some free port */
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port2 = rtp_session_get_local_port(audio_stream_->session);
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bool LinphoneVoiceChannel::SetSend(SendFlags flag) {
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void LinphoneVoiceChannel::OnPacketReceived(talk_base::Buffer* packet) {
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const void* data = packet->data();
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int len = packet->length();
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memcpy(buf, data, len);
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if (port2 == PORT_UNUSED)
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/* We may receive packets with payload type 13: comfort noise. Linphone can't
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* handle them, so let's ignore those packets.
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int payloadtype = buf[1] & 0x7f;
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if (play_ && payloadtype != 13)
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socket_->SendTo(buf, len, talk_base::SocketAddress("localhost",port2));
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void LinphoneVoiceChannel::StartRing(bool bIncomingCall)
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MSSndCard *sndcard = NULL;
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sndcard=ms_snd_card_manager_get_default_card(ms_snd_card_manager_get());
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if (engine_->GetRingWav().size() > 0)
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LOG(LS_VERBOSE) << "incoming ring. sound file: " << engine_->GetRingWav().c_str() << "\n";
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ring_stream_ = ring_start (engine_->GetRingWav().c_str(), 1, sndcard);
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if (engine_->GetCallWav().size() > 0)
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LOG(LS_VERBOSE) << "outgoing ring. sound file: " << engine_->GetCallWav().c_str() << "\n";
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ring_stream_ = ring_start (engine_->GetCallWav().c_str(), 1, sndcard);
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void LinphoneVoiceChannel::StopRing()
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ring_stop(ring_stream_);
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void LinphoneVoiceChannel::OnIncomingData(talk_base::AsyncSocket *s)
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len = s->Recv(buf, sizeof(buf));
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talk_base::Buffer packet(buf, len, sizeof(buf));
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if (network_interface_ && !mute_)
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network_interface_->SendPacket(&packet);
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#endif // HAVE_LINPHONE