245 lines
8.8 KiB
C++
245 lines
8.8 KiB
C++
/*
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* libjingle
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* Copyright 2004--2011, Google Inc.
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*
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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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*
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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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*
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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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*/
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#ifndef TALK_APP_WEBRTC_AUDIOMEDIAENGINE_H_
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#define TALK_APP_WEBRTC_AUDIOMEDIAENGINE_H_
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#include <map>
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#include <string>
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#include <vector>
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#include "talk/base/buffer.h"
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#include "talk/base/byteorder.h"
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#include "talk/base/logging.h"
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#include "talk/base/scoped_ptr.h"
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#include "talk/base/stream.h"
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#include "talk/session/phone/channel.h"
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#include "talk/session/phone/mediaengine.h"
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#include "talk/session/phone/rtputils.h"
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#include "talk/app/voiceengine.h"
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namespace cricket {
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class SoundclipMedia;
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class VoiceMediaChannel;
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}
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namespace webrtc {
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// MonitorStream is used to monitor a stream coming from WebRTC.
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// For now we just dump the data.
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class MonitorStream : public OutStream {
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virtual bool Write(const void *buf, int len) {
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return true;
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}
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};
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class AudioDeviceModule;
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class RtcVoiceMediaChannel;
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// RtcVoiceEngine is a class to be used with CompositeMediaEngine.
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// It uses the WebRTC VoiceEngine library for audio handling.
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class RtcVoiceEngine
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: public VoiceEngineObserver,
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public TraceCallback {
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public:
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RtcVoiceEngine(); // NOLINT
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// Dependency injection for testing.
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explicit RtcVoiceEngine(RtcWrapper* rtc_wrapper);
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~RtcVoiceEngine();
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bool Init();
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void Terminate();
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int GetCapabilities();
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cricket::VoiceMediaChannel* CreateChannel();
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cricket::SoundclipMedia* CreateSoundclip() { return NULL; }
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bool SetDevices(const cricket::Device* in_device,
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const cricket::Device* out_device);
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bool SetOptions(int options);
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bool GetOutputVolume(int* level);
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bool SetOutputVolume(int level);
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int GetInputLevel();
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bool SetLocalMonitor(bool enable);
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const std::vector<cricket::AudioCodec>& codecs();
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bool FindCodec(const cricket::AudioCodec& codec);
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bool FindRtcCodec(const cricket::AudioCodec& codec, CodecInst* gcodec);
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void SetLogging(int min_sev, const char* filter);
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// For tracking WebRTC channels. Needed because we have to pause them
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// all when switching devices.
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// May only be called by RtcVoiceMediaChannel.
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void RegisterChannel(RtcVoiceMediaChannel *channel);
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void UnregisterChannel(RtcVoiceMediaChannel *channel);
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RtcWrapper* webrtc() { return rtc_wrapper_.get(); }
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int GetLastRtcError();
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private:
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typedef std::vector<RtcVoiceMediaChannel *> ChannelList;
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struct CodecPref {
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const char* name;
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int clockrate;
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};
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void Construct();
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bool InitInternal();
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void ApplyLogging();
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virtual void Print(const TraceLevel level,
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const char* traceString, const int length);
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virtual void CallbackOnError(const int errCode, const int channel);
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static int GetCodecPreference(const char *name, int clockrate);
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// Given the device type, name, and id, find WebRTC's device id. Return true and
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// set the output parameter rtc_id if successful.
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bool FindAudioDeviceId(
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bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
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bool FindChannelAndSsrc(int channel_num,
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RtcVoiceMediaChannel** channel,
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uint32* ssrc) const;
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static const int kDefaultLogSeverity = talk_base::LS_WARNING;
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static const CodecPref kCodecPrefs[];
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// The primary instance of WebRTC VoiceEngine.
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talk_base::scoped_ptr<RtcWrapper> rtc_wrapper_;
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int log_level_;
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std::vector<cricket::AudioCodec> codecs_;
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talk_base::scoped_ptr<MonitorStream> monitor_;
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// TODO: Can't use scoped_ptr here since ~AudioDeviceModule is protected.
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AudioDeviceModule* adm_;
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ChannelList channels_;
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talk_base::CriticalSection channels_cs_;
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};
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// RtcMediaChannel is a class that implements the common WebRTC channel
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// functionality.
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template <class T, class E>
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class RtcMediaChannel : public T, public Transport {
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public:
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RtcMediaChannel(E *engine, int channel)
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: engine_(engine), audio_channel_(channel), sequence_number_(-1) {}
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E *engine() { return engine_; }
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int audio_channel() const { return audio_channel_; }
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bool valid() const { return audio_channel_ != -1; }
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protected:
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// implements Transport interface
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virtual int SendPacket(int channel, const void *data, int len) {
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if (!T::network_interface_) {
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return -1;
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}
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const uint8* header = static_cast<const uint8*>(data);
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sequence_number_ = talk_base::GetBE16(header + 2);
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talk_base::Buffer packet(data, len, cricket::kMaxRtpPacketLen);
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return T::network_interface_->SendPacket(&packet) ? len : -1;
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}
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virtual int SendRTCPPacket(int channel, const void *data, int len) {
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if (!T::network_interface_) {
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return -1;
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}
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talk_base::Buffer packet(data, len, cricket::kMaxRtpPacketLen);
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return T::network_interface_->SendRtcp(&packet) ? len : -1;
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}
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int sequence_number() {
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return sequence_number_;
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}
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private:
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E *engine_;
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int audio_channel_;
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int sequence_number_;
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};
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// RtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
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// WebRTC Voice Engine.
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class RtcVoiceMediaChannel
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: public RtcMediaChannel<cricket::VoiceMediaChannel, RtcVoiceEngine> {
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public:
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explicit RtcVoiceMediaChannel(RtcVoiceEngine *engine);
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virtual ~RtcVoiceMediaChannel();
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virtual bool SetOptions(int options);
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virtual bool SetRecvCodecs(const std::vector<cricket::AudioCodec> &codecs);
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virtual bool SetSendCodecs(const std::vector<cricket::AudioCodec> &codecs);
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virtual bool SetPlayout(bool playout);
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bool GetPlayout();
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virtual bool SetSend(cricket::SendFlags send);
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cricket::SendFlags GetSend();
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virtual bool AddStream(uint32 ssrc);
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virtual bool RemoveStream(uint32 ssrc);
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virtual bool GetActiveStreams(cricket::AudioInfo::StreamList* actives);
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virtual int GetOutputLevel();
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virtual bool SetRingbackTone(const char *buf, int len);
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virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop);
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virtual bool PlayRingbackTone(bool play, bool loop);
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virtual bool PressDTMF(int event, bool playout);
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virtual void OnPacketReceived(talk_base::Buffer* packet);
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virtual void OnRtcpReceived(talk_base::Buffer* packet);
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virtual void SetSendSsrc(uint32 id);
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virtual bool SetRtcpCName(const std::string& cname);
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virtual bool Mute(bool mute);
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virtual bool SetRecvRtpHeaderExtensions(
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const std::vector<cricket::RtpHeaderExtension>& extensions) { return false; }
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virtual bool SetSendRtpHeaderExtensions(
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const std::vector<cricket::RtpHeaderExtension>& extensions) { return false; }
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virtual bool SetSendBandwidth(bool autobw, int bps) { return false; }
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virtual bool GetStats(cricket::VoiceMediaInfo* info);
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virtual void GetLastMediaError(uint32* ssrc,
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VoiceMediaChannel::Error* error);
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bool FindSsrc(int channel_num, uint32* ssrc);
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void OnError(uint32 ssrc, int error);
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virtual int GetMediaChannelId() { return audio_channel(); }
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protected:
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int GetLastRtcError() { return engine()->GetLastRtcError(); }
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int GetChannel(uint32 ssrc);
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int GetOutputLevel(int channel);
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bool EnableRtcp(int channel);
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bool SetPlayout(int channel, bool playout);
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static uint32 ParseSsrc(const void* data, size_t len, bool rtcp);
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static Error WebRTCErrorToChannelError(int err_code);
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private:
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typedef std::map<uint32, int> ChannelMap;
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int channel_options_;
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bool playout_;
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cricket::SendFlags send_;
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ChannelMap mux_channels_; // for multiple sources
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// mux_channels_ can be read from WebRTC callback thread. Accesses off the
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// WebRTC thread must be synchronized with edits on the worker thread. Reads
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// on the worker thread are ok.
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mutable talk_base::CriticalSection mux_channels_cs_;
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};
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} // namespace webrtc
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#endif // TALK_APP_WEBRTC_AUDIOMEDIAENGINE_H_
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