// Copyright 2019 Fuzhou Rockchip Electronics Co., Ltd. All rights reserved.
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// Use of this source code is governed by a BSD-style license that can be
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// found in the LICENSE file.
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#include "aac_server_media_subsession.hh"
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#include "MPEG4GenericRTPSink.hh"
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#include "utils.h"
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namespace easymedia
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{
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static unsigned const samplingFrequencyTable[16] = {96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
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16000, 12000, 11025, 8000, 7350, 0, 0, 0};
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AACServerMediaSubsession* AACServerMediaSubsession::createNew(UsageEnvironment& env, Live555MediaInput& wisInput,
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unsigned samplingFrequency, unsigned numChannels,
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unsigned char profile)
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{
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return new AACServerMediaSubsession(env, wisInput, samplingFrequency, numChannels, profile);
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}
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AACServerMediaSubsession::AACServerMediaSubsession(UsageEnvironment& env, Live555MediaInput& mediaInput,
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unsigned samplingFrequency, unsigned numChannels,
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unsigned char profile)
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: OnDemandServerMediaSubsession(env, True /*reuse the first source*/), fMediaInput(mediaInput),
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fSamplingFrequency(samplingFrequency), fNumChannels(numChannels)
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{
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unsigned char audioSpecificConfig[2];
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unsigned char const audioObjectType = profile + 1;
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unsigned char samplingFrequencyIndex = 0;
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for (unsigned char i = 0; i < 16; i++) {
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if (samplingFrequencyTable[i] == fSamplingFrequency) {
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samplingFrequencyIndex = i;
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break;
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}
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}
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audioSpecificConfig[0] = (audioObjectType << 3) | (samplingFrequencyIndex >> 1);
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// audioSpecificConfig[1] = (samplingFrequencyIndex<<7) |
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// (channelConfiguration<<3);
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audioSpecificConfig[1] = (samplingFrequencyIndex << 7) | (fNumChannels << 3);
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sprintf(fConfigStr, "%02X%02x", audioSpecificConfig[0], audioSpecificConfig[1]);
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}
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AACServerMediaSubsession::~AACServerMediaSubsession()
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{
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LOG_FILE_FUNC_LINE();
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}
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FramedSource* AACServerMediaSubsession::createNewStreamSource(unsigned /*clientSessionId*/, unsigned& estBitrate)
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{
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LOG_FILE_FUNC_LINE();
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estBitrate = 96; // kbps, estimate
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return fMediaInput.audioSource();
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}
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RTPSink* AACServerMediaSubsession::createNewRTPSink(Groupsock* rtpGroupsock, unsigned char rtpPayloadTypeIfDynamic,
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FramedSource* inputSource)
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{
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if (!inputSource) {
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LOG("inputSource is not ready, can not create new rtp sink\n");
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return NULL;
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}
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// ADTSAudioFileSource* adtsSource = (ADTSAudioFileSource*)inputSource;
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setAudioRTPSinkBufferSize();
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RTPSink* rtpsink =
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MPEG4GenericRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic, fSamplingFrequency, "audio",
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"AAC-hbr", fConfigStr, fNumChannels);
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setVideoRTPSinkBufferSize();
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return rtpsink;
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}
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// std::mutex AACServerMediaSubsession::kMutex;
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void AACServerMediaSubsession::startStream(unsigned clientSessionId, void* streamToken, TaskFunc* rtcpRRHandler,
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void* rtcpRRHandlerClientData, unsigned short& rtpSeqNum,
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unsigned& rtpTimestamp,
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ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
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void* serverRequestAlternativeByteHandlerClientData)
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{
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OnDemandServerMediaSubsession::startStream(clientSessionId, streamToken, rtcpRRHandler, rtcpRRHandlerClientData,
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rtpSeqNum, rtpTimestamp, serverRequestAlternativeByteHandler,
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serverRequestAlternativeByteHandlerClientData);
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// kMutex.lock();
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if (kSessionIdList.empty()) {
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fMediaInput.Start(envir());
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}
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if (fMediaInput.GetStartAudioStreamCallback() != NULL) {
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fMediaInput.GetStartAudioStreamCallback()();
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}
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LOG("%s:%s:%p - clientSessionId: 0x%08x\n", __FILE__, __func__, this, clientSessionId);
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kSessionIdList.push_back(clientSessionId);
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// kMutex.unlock();
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}
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void AACServerMediaSubsession::deleteStream(unsigned clientSessionId, void*& streamToken)
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{
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// kMutex.lock();
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LOG("%s - clientSessionId: 0x%08x\n", __func__, clientSessionId);
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kSessionIdList.remove(clientSessionId);
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if (kSessionIdList.empty()) {
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fMediaInput.Stop(envir());
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}
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// kMutex.unlock();
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OnDemandServerMediaSubsession::deleteStream(clientSessionId, streamToken);
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}
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} // namespace easymedia
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