void CodecFactory::removeCodec(const std::string& codec_id) {
// convert PluginNotFound exceptions into CodecNotFound exceptions
try {
PluginConfig<Codec>::removePlugin(codec_id);
} catch (PluginManager<Codec>::PluginNotFoundException&) {
throw CodecNotFoundException(codec_id);
}
}
void CodecFactory::setCodecConfig(const std::string& codec_id,
const xmlNodePtr config_ptr)
{
// convert PluginNotFound exceptions into CodecNotFound exceptions
try {
PluginConfig<Codec>::setPluginConfig(codec_id, config_ptr);
} catch (PluginManager<Codec>::PluginNotFoundException&) {
throw CodecNotFoundException(codec_id);
}
}
CodecPtr CodecFactory::getCodec(const std::string& codec_id)
{
boost::mutex::scoped_lock factory_lock(m_mutex);
Codec *codec_ptr = m_plugins.get(codec_id);
// throw an exception if the codec was not found
if (codec_ptr == NULL)
throw CodecNotFoundException(codec_id);
// return a cloned instance of the Codec since its state may change
// while encoding or decoding data streams
return codec_ptr->clone();
}
FFData(const std::string &path, VSFileSystem::VSFileType type, Format &fmt, int streamIdx) throw(Exception) :
pFormatCtx(0),
pCodecCtx(0),
pCodec(0),
pStream(0),
packetBuffer(0),
packetBufferSize(0),
sampleBufferBase(0),
filepath(path),
filetype(type),
audioStreamIndex(streamIdx)
{
packet.data = 0;
char buf[(sizeof(type)+1)/2+1];
sprintf(buf, "%d", type);
// Initialize libavcodec/libavformat if necessary
FFMpeg::initLibraries();
// Open file
std::string npath = std::string("vsfile:") + path + "|" + buf;
std::string errbase = std::string("Cannot open URL \"") + npath + "\"";
if ( (0 != av_open_input_file(&pFormatCtx, npath.c_str(), NULL, BUFFER_SIZE, NULL))
||(0 > av_find_stream_info(pFormatCtx)) )
throw FileOpenException(errbase + " (wrong format or file not found)");
// Dump format info in case we want to know...
#ifdef VS_DEBUG
dump_format(pFormatCtx, 0, npath.c_str(), false);
#endif
// Find audio stream
pCodecCtx = 0;
streamIndex = -1;
for (unsigned int i=0; (pCodecCtx==0) && (i < pFormatCtx->nb_streams); ++i)
if ((pFormatCtx->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO) && (streamIdx-- == 0))
pCodecCtx = (pStream = pFormatCtx->streams[streamIndex = i])->codec;
if (pCodecCtx == 0)
throw FileOpenException(errbase + " (wrong or no audio stream)");
// Find codec for the audio stream and open it
pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
if(pCodec == 0)
throw CodecNotFoundException(errbase + " (unsupported codec)");
if(avcodec_open(pCodecCtx, pCodec) < 0)
throw CodecNotFoundException(errbase + " (unsupported codec)");
// Get some info
fmt.sampleFrequency = pCodecCtx->sample_rate;
fmt.channels = pCodecCtx->channels;
fmt.nativeOrder = 1; // always so for ffmpeg
switch (pCodecCtx->sample_fmt) {
case SAMPLE_FMT_U8: fmt.bitsPerSample = 8;
fmt.signedSamples = 0;
break;
case SAMPLE_FMT_S16: fmt.bitsPerSample = 16;
fmt.signedSamples = 1;
break;
#ifdef SAMPLE_FMT_S24
case SAMPLE_FMT_S24: fmt.bitsPerSample = 24;
fmt.signedSamples = 1;
break;
#endif
#ifdef SAMPLE_FMT_S32
case SAMPLE_FMT_S32: fmt.bitsPerSample = 32;
fmt.signedSamples = 1;
break;
#endif
default: throw CodecNotFoundException(errbase + " (unsupported audio format)");
}
sampleSize = (fmt.bitsPerSample + 7) / 8 * fmt.channels;
assert(sampleSize > 0);
// Initialize timebase counter
sampleBufferStart = 0;
streamSize = 0;
// Initialize sample buffer
sampleBufferBase = malloc(sampleSize * BUFFER_SIZE + BUFFER_ALIGNMENT);
ptrdiff_t offs = ((reinterpret_cast<ptrdiff_t>(sampleBufferBase)) & (BUFFER_ALIGNMENT-1));
sampleBufferAligned = ((char*)sampleBufferBase) + BUFFER_ALIGNMENT - offs;
sampleBufferAlloc = sampleSize * BUFFER_SIZE;
sampleBuffer = 0;
sampleBufferSize = 0;
}
