1.1 --- /dev/null Thu Jan 01 00:00:00 1970 +0000
1.2 +++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.h Wed Dec 31 06:09:35 2014 +0100
1.3 @@ -0,0 +1,275 @@
1.4 +/* This Source Code Form is subject to the terms of the Mozilla Public
1.5 + * License, v. 2.0. If a copy of the MPL was not distributed with this file,
1.6 + * You can obtain one at http://mozilla.org/MPL/2.0/. */
1.7 +
1.8 +
1.9 +#ifndef AUDIO_SESSION_H_
1.10 +#define AUDIO_SESSION_H_
1.11 +
1.12 +#include "mozilla/Attributes.h"
1.13 +#include "mozilla/TimeStamp.h"
1.14 +#include "nsTArray.h"
1.15 +
1.16 +#include "MediaConduitInterface.h"
1.17 +#include "MediaEngineWrapper.h"
1.18 +
1.19 +// Audio Engine Includes
1.20 +#include "webrtc/common_types.h"
1.21 +#include "webrtc/voice_engine/include/voe_base.h"
1.22 +#include "webrtc/voice_engine/include/voe_volume_control.h"
1.23 +#include "webrtc/voice_engine/include/voe_codec.h"
1.24 +#include "webrtc/voice_engine/include/voe_file.h"
1.25 +#include "webrtc/voice_engine/include/voe_network.h"
1.26 +#include "webrtc/voice_engine/include/voe_external_media.h"
1.27 +#include "webrtc/voice_engine/include/voe_audio_processing.h"
1.28 +#include "webrtc/voice_engine/include/voe_video_sync.h"
1.29 +#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
1.30 +//Some WebRTC types for short notations
1.31 + using webrtc::VoEBase;
1.32 + using webrtc::VoENetwork;
1.33 + using webrtc::VoECodec;
1.34 + using webrtc::VoEExternalMedia;
1.35 + using webrtc::VoEAudioProcessing;
1.36 + using webrtc::VoEVideoSync;
1.37 + using webrtc::VoERTP_RTCP;
1.38 +/** This file hosts several structures identifying different aspects
1.39 + * of a RTP Session.
1.40 + */
1.41 +namespace mozilla {
1.42 +// Helper function
1.43 +
1.44 +DOMHighResTimeStamp
1.45 +NTPtoDOMHighResTimeStamp(uint32_t ntpHigh, uint32_t ntpLow);
1.46 +
1.47 +/**
1.48 + * Concrete class for Audio session. Hooks up
1.49 + * - media-source and target to external transport
1.50 + */
1.51 +class WebrtcAudioConduit:public AudioSessionConduit
1.52 + ,public webrtc::Transport
1.53 +{
1.54 +public:
1.55 + //VoiceEngine defined constant for Payload Name Size.
1.56 + static const unsigned int CODEC_PLNAME_SIZE;
1.57 +
1.58 + /**
1.59 + * APIs used by the registered external transport to this Conduit to
1.60 + * feed in received RTP Frames to the VoiceEngine for decoding
1.61 + */
1.62 + virtual MediaConduitErrorCode ReceivedRTPPacket(const void *data, int len);
1.63 +
1.64 + /**
1.65 + * APIs used by the registered external transport to this Conduit to
1.66 + * feed in received RTCP Frames to the VoiceEngine for decoding
1.67 + */
1.68 + virtual MediaConduitErrorCode ReceivedRTCPPacket(const void *data, int len);
1.69 +
1.70 + /**
1.71 + * Function to configure send codec for the audio session
1.72 + * @param sendSessionConfig: CodecConfiguration
1.73 + * @result: On Success, the audio engine is configured with passed in codec for send
1.74 + * On failure, audio engine transmit functionality is disabled.
1.75 + * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
1.76 + * transmission sub-system on the engine.
1.77 + */
1.78 + virtual MediaConduitErrorCode ConfigureSendMediaCodec(const AudioCodecConfig* codecConfig);
1.79 + /**
1.80 + * Function to configure list of receive codecs for the audio session
1.81 + * @param sendSessionConfig: CodecConfiguration
1.82 + * @result: On Success, the audio engine is configured with passed in codec for send
1.83 + * Also the playout is enabled.
1.84 + * On failure, audio engine transmit functionality is disabled.
1.85 + * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
1.86 + * transmission sub-system on the engine.
1.87 + */
1.88 + virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
1.89 + const std::vector<AudioCodecConfig* >& codecConfigList);
1.90 + /**
1.91 + * Function to enable the audio level extension
1.92 + * @param enabled: enable extension
1.93 + */
1.94 + virtual MediaConduitErrorCode EnableAudioLevelExtension(bool enabled, uint8_t id);
1.95 +
1.96 + /**
1.97 + * Register External Transport to this Conduit. RTP and RTCP frames from the VoiceEngine
1.98 + * shall be passed to the registered transport for transporting externally.
1.99 + */
1.100 + virtual MediaConduitErrorCode AttachTransport(mozilla::RefPtr<TransportInterface> aTransport);
1.101 + /**
1.102 + * Function to deliver externally captured audio sample for encoding and transport
1.103 + * @param audioData [in]: Pointer to array containing a frame of audio
1.104 + * @param lengthSamples [in]: Length of audio frame in samples in multiple of 10 milliseconds
1.105 + * Ex: Frame length is 160, 320, 440 for 16, 32, 44 kHz sampling rates
1.106 + respectively.
1.107 + audioData[] should be of lengthSamples in size
1.108 + say, for 16kz sampling rate, audioData[] should contain 160
1.109 + samples of 16-bits each for a 10m audio frame.
1.110 + * @param samplingFreqHz [in]: Frequency/rate of the sampling in Hz ( 16000, 32000 ...)
1.111 + * @param capture_delay [in]: Approx Delay from recording until it is delivered to VoiceEngine
1.112 + in milliseconds.
1.113 + * NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can be invoked
1.114 + * This ensures the inserted audio-samples can be transmitted by the conduit
1.115 + *
1.116 + */
1.117 + virtual MediaConduitErrorCode SendAudioFrame(const int16_t speechData[],
1.118 + int32_t lengthSamples,
1.119 + int32_t samplingFreqHz,
1.120 + int32_t capture_time);
1.121 +
1.122 + /**
1.123 + * Function to grab a decoded audio-sample from the media engine for rendering
1.124 + * / playoutof length 10 milliseconds.
1.125 + *
1.126 + * @param speechData [in]: Pointer to a array to which a 10ms frame of audio will be copied
1.127 + * @param samplingFreqHz [in]: Frequency of the sampling for playback in Hertz (16000, 32000,..)
1.128 + * @param capture_delay [in]: Estimated Time between reading of the samples to rendering/playback
1.129 + * @param lengthSamples [out]: Will contain length of the audio frame in samples at return.
1.130 + Ex: A value of 160 implies 160 samples each of 16-bits was copied
1.131 + into speechData
1.132 + * NOTE: This function should be invoked every 10 milliseconds for the best
1.133 + * peformance
1.134 + * NOTE: ConfigureRecvMediaCodec() SHOULD be called before this function can be invoked
1.135 + * This ensures the decoded samples are ready for reading and playout is enabled.
1.136 + *
1.137 + */
1.138 + virtual MediaConduitErrorCode GetAudioFrame(int16_t speechData[],
1.139 + int32_t samplingFreqHz,
1.140 + int32_t capture_delay,
1.141 + int& lengthSamples);
1.142 +
1.143 +
1.144 + /**
1.145 + * Webrtc transport implementation to send and receive RTP packet.
1.146 + * AudioConduit registers itself as ExternalTransport to the VoiceEngine
1.147 + */
1.148 + virtual int SendPacket(int channel, const void *data, int len) ;
1.149 +
1.150 + /**
1.151 + * Webrtc transport implementation to send and receive RTCP packet.
1.152 + * AudioConduit registers itself as ExternalTransport to the VoiceEngine
1.153 + */
1.154 + virtual int SendRTCPPacket(int channel, const void *data, int len) ;
1.155 +
1.156 +
1.157 +
1.158 + WebrtcAudioConduit():
1.159 + mOtherDirection(nullptr),
1.160 + mShutDown(false),
1.161 + mVoiceEngine(nullptr),
1.162 + mTransport(nullptr),
1.163 + mEngineTransmitting(false),
1.164 + mEngineReceiving(false),
1.165 + mChannel(-1),
1.166 + mCurSendCodecConfig(nullptr),
1.167 + mCaptureDelay(150),
1.168 +#ifdef MOZILLA_INTERNAL_API
1.169 + mLastTimestamp(0),
1.170 +#endif // MOZILLA_INTERNAL_API
1.171 + mSamples(0),
1.172 + mLastSyncLog(0)
1.173 + {
1.174 + }
1.175 +
1.176 + virtual ~WebrtcAudioConduit();
1.177 +
1.178 + MediaConduitErrorCode Init(WebrtcAudioConduit *other);
1.179 +
1.180 + int GetChannel() { return mChannel; }
1.181 + webrtc::VoiceEngine* GetVoiceEngine() { return mVoiceEngine; }
1.182 + bool GetLocalSSRC(unsigned int* ssrc);
1.183 + bool GetRemoteSSRC(unsigned int* ssrc);
1.184 + bool GetAVStats(int32_t* jitterBufferDelayMs,
1.185 + int32_t* playoutBufferDelayMs,
1.186 + int32_t* avSyncOffsetMs);
1.187 + bool GetRTPStats(unsigned int* jitterMs, unsigned int* cumulativeLost);
1.188 + bool GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
1.189 + uint32_t* jitterMs,
1.190 + uint32_t* packetsReceived,
1.191 + uint64_t* bytesReceived,
1.192 + uint32_t *cumulativeLost,
1.193 + int32_t* rttMs);
1.194 + bool GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
1.195 + unsigned int* packetsSent,
1.196 + uint64_t* bytesSent);
1.197 +
1.198 +private:
1.199 + WebrtcAudioConduit(const WebrtcAudioConduit& other) MOZ_DELETE;
1.200 + void operator=(const WebrtcAudioConduit& other) MOZ_DELETE;
1.201 +
1.202 + //Local database of currently applied receive codecs
1.203 + typedef std::vector<AudioCodecConfig* > RecvCodecList;
1.204 +
1.205 + //Function to convert between WebRTC and Conduit codec structures
1.206 + bool CodecConfigToWebRTCCodec(const AudioCodecConfig* codecInfo,
1.207 + webrtc::CodecInst& cinst);
1.208 +
1.209 + //Checks if given sampling frequency is supported
1.210 + bool IsSamplingFreqSupported(int freq) const;
1.211 +
1.212 + //Generate block size in sample lenght for a given sampling frequency
1.213 + unsigned int GetNum10msSamplesForFrequency(int samplingFreqHz) const;
1.214 +
1.215 + // Function to copy a codec structure to Conduit's database
1.216 + bool CopyCodecToDB(const AudioCodecConfig* codecInfo);
1.217 +
1.218 + // Functions to verify if the codec passed is already in
1.219 + // conduits database
1.220 + bool CheckCodecForMatch(const AudioCodecConfig* codecInfo) const;
1.221 + bool CheckCodecsForMatch(const AudioCodecConfig* curCodecConfig,
1.222 + const AudioCodecConfig* codecInfo) const;
1.223 + //Checks the codec to be applied
1.224 + MediaConduitErrorCode ValidateCodecConfig(const AudioCodecConfig* codecInfo, bool send) const;
1.225 +
1.226 + //Utility function to dump recv codec database
1.227 + void DumpCodecDB() const;
1.228 +
1.229 + // The two sides of a send/receive pair of conduits each keep a pointer to the other.
1.230 + // The also share a single VoiceEngine and mChannel. Shutdown must be coordinated
1.231 + // carefully to avoid double-freeing or accessing after one frees.
1.232 + WebrtcAudioConduit* mOtherDirection;
1.233 + // Other side has shut down our channel and related items already
1.234 + bool mShutDown;
1.235 +
1.236 + // These are shared by both directions. They're released by the last
1.237 + // conduit to die
1.238 + webrtc::VoiceEngine* mVoiceEngine;
1.239 + mozilla::RefPtr<TransportInterface> mTransport;
1.240 + ScopedCustomReleasePtr<webrtc::VoENetwork> mPtrVoENetwork;
1.241 + ScopedCustomReleasePtr<webrtc::VoEBase> mPtrVoEBase;
1.242 + ScopedCustomReleasePtr<webrtc::VoECodec> mPtrVoECodec;
1.243 + ScopedCustomReleasePtr<webrtc::VoEExternalMedia> mPtrVoEXmedia;
1.244 + ScopedCustomReleasePtr<webrtc::VoEAudioProcessing> mPtrVoEProcessing;
1.245 + ScopedCustomReleasePtr<webrtc::VoEVideoSync> mPtrVoEVideoSync;
1.246 + ScopedCustomReleasePtr<webrtc::VoERTP_RTCP> mPtrVoERTP_RTCP;
1.247 + ScopedCustomReleasePtr<webrtc::VoERTP_RTCP> mPtrRTP;
1.248 + //engine states of our interets
1.249 + bool mEngineTransmitting; // If true => VoiceEngine Send-subsystem is up
1.250 + bool mEngineReceiving; // If true => VoiceEngine Receive-subsystem is up
1.251 + // and playout is enabled
1.252 + // Keep track of each inserted RTP block and the time it was inserted
1.253 + // so we can estimate the clock time for a specific TimeStamp coming out
1.254 + // (for when we send data to MediaStreamTracks). Blocks are aged out as needed.
1.255 + struct Processing {
1.256 + TimeStamp mTimeStamp;
1.257 + uint32_t mRTPTimeStamp; // RTP timestamps received
1.258 + };
1.259 + nsAutoTArray<Processing,8> mProcessing;
1.260 +
1.261 + int mChannel;
1.262 + RecvCodecList mRecvCodecList;
1.263 + AudioCodecConfig* mCurSendCodecConfig;
1.264 +
1.265 + // Current "capture" delay (really output plus input delay)
1.266 + int32_t mCaptureDelay;
1.267 +
1.268 +#ifdef MOZILLA_INTERNAL_API
1.269 + uint32_t mLastTimestamp;
1.270 +#endif // MOZILLA_INTERNAL_API
1.271 +
1.272 + uint32_t mSamples;
1.273 + uint32_t mLastSyncLog;
1.274 +};
1.275 +
1.276 +} // end namespace
1.277 +
1.278 +#endif
