IAudioSource* cAudioManager::createFromMemory(const char* name, const char* data, size_t length, const char* extension)
{
if(!Initialized) return NULL;
cAudioMutexBasicLock lock(Mutex);
cAudioString audioName = safeCStr(name);
cAudioString ext = safeCStr(extension);
IAudioDecoderFactory* factory = getAudioDecoderFactory(ext.c_str());
if(!factory) {
getLogger()->logError("AudioManager", "Failed to create Audio Source (%s): Codec (.%s) is not supported.", audioName.c_str(), ext.c_str());
return NULL;
}
cMemorySource* source = CAUDIO_NEW cMemorySource(data, length, true);
if(source && source->isValid())
{
IAudioDecoder* decoder = factory->CreateAudioDecoder(source);
source->drop();
IAudioSource* audio = createAudioSource(decoder, audioName, "cMemorySource");
if(audio != NULL)
return audio;
if(source)
source->drop();
}
return NULL;
}
IAudioSource* cAudioManager::createFromRaw(const char* name, const char* data, size_t length, unsigned int frequency, AudioFormats format)
{
if(!Initialized) return NULL;
cAudioMutexBasicLock lock(Mutex);
cAudioString audioName = safeCStr(name);
IAudioDecoderFactory* factory = getAudioDecoderFactory("raw");
if(!factory) {
getLogger()->logError("AudioManager", "Failed to create Audio Source (%s): Codec (.raw) is not supported.", audioName.c_str());
return NULL;
}
cMemorySource* source = CAUDIO_NEW cMemorySource(data, length, true);
if(source && source->isValid())
{
IAudioDecoder* decoder = ((cRawAudioDecoderFactory*)factory)->CreateAudioDecoder(source, frequency, format);
source->drop();
IAudioSource* audio = createAudioSource(decoder, audioName, "cMemorySource");
if(audio != NULL)
return audio;
if(source)
source->drop();
}
return NULL;
}
status_t StagefrightRecorder::startRTPRecording() {
CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_RTP_AVP);
if ((mAudioSource != AUDIO_SOURCE_LIST_END
&& mVideoSource != VIDEO_SOURCE_LIST_END)
|| (mAudioSource == AUDIO_SOURCE_LIST_END
&& mVideoSource == VIDEO_SOURCE_LIST_END)) {
// Must have exactly one source.
return BAD_VALUE;
}
if (mOutputFd < 0) {
return BAD_VALUE;
}
sp<MediaSource> source;
if (mAudioSource != AUDIO_SOURCE_LIST_END) {
source = createAudioSource();
} else {
status_t err = setupVideoEncoder(&source);
if (err != OK) {
return err;
}
}
mWriter = new ARTPWriter(dup(mOutputFd));
mWriter->addSource(source);
mWriter->setListener(mListener);
return mWriter->start();
}
status_t StagefrightRecorder::startAMRRecording() {
CHECK(mOutputFormat == OUTPUT_FORMAT_AMR_NB ||
mOutputFormat == OUTPUT_FORMAT_AMR_WB);
if (mOutputFormat == OUTPUT_FORMAT_AMR_NB) {
if (mAudioEncoder != AUDIO_ENCODER_DEFAULT &&
mAudioEncoder != AUDIO_ENCODER_AMR_NB) {
LOGE("Invalid encoder %d used for AMRNB recording",
mAudioEncoder);
return BAD_VALUE;
}
if (mSampleRate != 8000) {
LOGE("Invalid sampling rate %d used for AMRNB recording",
mSampleRate);
return BAD_VALUE;
}
} else { // mOutputFormat must be OUTPUT_FORMAT_AMR_WB
if (mAudioEncoder != AUDIO_ENCODER_AMR_WB) {
LOGE("Invlaid encoder %d used for AMRWB recording",
mAudioEncoder);
return BAD_VALUE;
}
if (mSampleRate != 16000) {
LOGE("Invalid sample rate %d used for AMRWB recording",
mSampleRate);
return BAD_VALUE;
}
}
if (mAudioChannels != 1) {
LOGE("Invalid number of audio channels %d used for amr recording",
mAudioChannels);
return BAD_VALUE;
}
if (mAudioSource >= AUDIO_SOURCE_LIST_END) {
LOGE("Invalid audio source: %d", mAudioSource);
return BAD_VALUE;
}
sp<MediaSource> audioEncoder = createAudioSource();
if (audioEncoder == NULL) {
return UNKNOWN_ERROR;
}
mWriter = new AMRWriter(dup(mOutputFd));
mWriter->addSource(audioEncoder);
if (mMaxFileDurationUs != 0) {
mWriter->setMaxFileDuration(mMaxFileDurationUs);
}
if (mMaxFileSizeBytes != 0) {
mWriter->setMaxFileSize(mMaxFileSizeBytes);
}
mWriter->setListener(mListener);
mWriter->start();
return OK;
}
status_t StagefrightRecorder::setupAudioEncoder(const sp<MediaWriter>& writer) {
sp<MediaSource> audioEncoder;
switch(mAudioEncoder) {
case AUDIO_ENCODER_AMR_NB:
case AUDIO_ENCODER_AMR_WB:
case AUDIO_ENCODER_AAC:
audioEncoder = createAudioSource();
break;
default:
LOGE("Unsupported audio encoder: %d", mAudioEncoder);
return UNKNOWN_ERROR;
}
if (audioEncoder == NULL) {
return UNKNOWN_ERROR;
}
writer->addSource(audioEncoder);
return OK;
}
IAudioSource* cAudioManager::create(const char* name, const char* filename, bool stream)
{
if(!Initialized) return NULL;
cAudioMutexBasicLock lock(Mutex);
cAudioString audioName = safeCStr(name);
cAudioString path = safeCStr(filename);
cAudioString ext = getExt(path);
IAudioDecoderFactory* factory = getAudioDecoderFactory(ext.c_str());
if(!factory) {
getLogger()->logError("AudioManager", "Failed to create Audio Source (%s): No decoder could be found for (.%s).", audioName.c_str(), ext.c_str());
return NULL;
}
for(size_t i=0; i<dataSourcePriorityList.size(); ++i)
{
const cAudioString dataSourceName = dataSourcePriorityList[i].second;
IDataSourceFactory* dataFactory = datasourcemap[dataSourceName];
if(dataFactory)
{
IDataSource* source = dataFactory->CreateDataSource(filename, stream);
if(source && source->isValid())
{
IAudioDecoder* decoder = factory->CreateAudioDecoder(source);
source->drop();
IAudioSource* audio = createAudioSource(decoder, audioName, dataSourceName);
if(audio != NULL)
return audio;
if(source)
source->drop();
return NULL;
}
}
}
return NULL;
}
void setupDarwinStreaming(UsageEnvironment& env, WISInput& inputDevice) {
// Create a 'Darwin injector' object:
injector = DarwinInjector::createNew(env, applicationName);
// For RTCP:
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen + 1];
gethostname((char *) CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
/******************audio***********************/
if (audioFormat != AFMT_NONE) {
// Create the audio source:
sourceAudio = createAudioSource(env, inputDevice.audioSource());
if (packageFormat != PFMT_TRANSPORT_STREAM) { // there's a separate RTP stream for audio
// Create 'groupsocks' for RTP and RTCP.
// (Note: Because we will actually be streaming through a remote Darwin server,
// via TCP, we just use dummy destination addresses, port numbers, and TTLs here.)
struct in_addr dummyDestAddress;
dummyDestAddress.s_addr = 0;
rtpGroupsockAudio = new Groupsock(env, dummyDestAddress, 0, 0);
rtcpGroupsockAudio = new Groupsock(env, dummyDestAddress, 0, 0);
// Create a RTP sink for the audio stream:
sinkAudio = createAudioRTPSink(env, rtpGroupsockAudio);
// Create (and start) a 'RTCP instance' for this RTP sink:
unsigned totalSessionBandwidthAudio = (audioOutputBitrate+500)/1000; // in kbps; for RTCP b/w share
rtcpAudio = RTCPInstance::createNew(env, rtcpGroupsockAudio,
totalSessionBandwidthAudio, CNAME,
sinkAudio, NULL /* we're a server */);
// Note: This starts RTCP running automatically
// Add these to our 'Darwin injector':
injector->addStream(sinkAudio, rtcpAudio);
}
}
/******************end audio***********************/
/******************video***********************/
if (videoFormat != VFMT_NONE) {
// Create the video source:
if (packageFormat == PFMT_TRANSPORT_STREAM) {
MPEG2TransportStreamFromESSource* tsSource
= MPEG2TransportStreamFromESSource::createNew(env);
tsSource->addNewVideoSource(inputDevice.videoSource(), 2);
if (sourceAudio != NULL) tsSource->addNewAudioSource(sourceAudio, 2);
// Gather the Transport packets into network packet-sized chunks:
sourceVideo = MPEG2TransportStreamAccumulator::createNew(env, tsSource);
sourceAudio = NULL;
} else {
switch (videoFormat) {
case VFMT_NONE: // not used
break;
case VFMT_MJPEG: {
sourceVideo = WISJPEGStreamSource::createNew(inputDevice.videoSource());
break;
}
case VFMT_MPEG1:
case VFMT_MPEG2: {
sourceVideo = MPEG1or2VideoStreamDiscreteFramer::createNew(env, inputDevice.videoSource());
break;
}
case VFMT_MPEG4: {
sourceVideo = MPEG4VideoStreamDiscreteFramer::createNew(env, inputDevice.videoSource());
break;
}
}
}
// Create 'groupsocks' for RTP and RTCP.
// (Note: Because we will actually be streaming through a remote Darwin server,
// via TCP, we just use dummy destination addresses, port numbers, and TTLs here.)
struct in_addr dummyDestAddress;
dummyDestAddress.s_addr = 0;
rtpGroupsockVideo = new Groupsock(env, dummyDestAddress, 0, 0);
rtcpGroupsockVideo = new Groupsock(env, dummyDestAddress, 0, 0);
// Create a RTP sink for the video stream:
unsigned char payloadFormatCode = 97; // if dynamic
setVideoRTPSinkBufferSize();
if (packageFormat == PFMT_TRANSPORT_STREAM) {
sinkVideo = SimpleRTPSink::createNew(env, rtpGroupsockVideo,
33, 90000, "video", "mp2t",
1, True, False/*no 'M' bit*/);
} else {
switch (videoFormat) {
case VFMT_NONE: // not used
break;
case VFMT_MJPEG: {
sinkVideo = JPEGVideoRTPSink::createNew(env, rtpGroupsockVideo);
break;
}
case VFMT_MPEG1:
case VFMT_MPEG2: {
sinkVideo = MPEG1or2VideoRTPSink::createNew(env, rtpGroupsockVideo);
break;
}
case VFMT_MPEG4: {
sinkVideo = MPEG4ESVideoRTPSink::createNew(env, rtpGroupsockVideo, payloadFormatCode);
break;
}
}
}
// Create (and start) a 'RTCP instance' for this RTP sink:
unsigned totalSessionBandwidthVideo = (videoBitrate+500)/1000; // in kbps; for RTCP b/w share
rtcpVideo = RTCPInstance::createNew(env, rtcpGroupsockVideo,
totalSessionBandwidthVideo, CNAME,
sinkVideo, NULL /* we're a server */);
// Note: This starts RTCP running automatically
// Add these to our 'Darwin injector':
injector->addStream(sinkVideo, rtcpVideo);
}
/******************end video***********************/
// Next, specify the destination Darwin Streaming Server:
char const* remoteStreamName = "test.sdp";//#####@@@@@
if (!injector->setDestination(remoteDSSNameOrAddress, remoteStreamName,
applicationName, "LIVE555 Streaming Media")) {
env << "Failed to connect to remote Darwin Streaming Server: " << env.getResultMsg() << "\n";
exit(1);
}
env << "Play this stream (from the Darwin Streaming Server) using the URL:\n"
<< "\trtsp://" << remoteDSSNameOrAddress << "/" << remoteStreamName << "\n";
}
