本文整理汇总了C++中nsRefPtr::BreakCycles方法的典型用法代码示例。如果您正苦于以下问题：C++ nsRefPtr::BreakCycles方法的具体用法？C++ nsRefPtr::BreakCycles怎么用？C++ nsRefPtr::BreakCycles使用的例子？那么恭喜您, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在类nsRefPtr的用法示例。
在下文中一共展示了nsRefPtr::BreakCycles方法的1个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于系统推荐出更棒的C++代码示例。
示例1: ReportFailureOnMainThread
void
MediaDecodeTask::Decode()
{
MOZ_ASSERT(!NS_IsMainThread());
mBufferDecoder->BeginDecoding(NS_GetCurrentThread());
// Tell the decoder reader that we are not going to play the data directly,
// and that we should not reject files with more channels than the audio
// bakend support.
mDecoderReader->SetIgnoreAudioOutputFormat();
MediaInfo mediaInfo;
nsAutoPtr<MetadataTags> tags;
nsresult rv = mDecoderReader->ReadMetadata(&mediaInfo, getter_Transfers(tags));
if (NS_FAILED(rv)) {
ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
return;
}
if (!mDecoderReader->HasAudio()) {
ReportFailureOnMainThread(WebAudioDecodeJob::NoAudio);
return;
}
MediaQueue<AudioData> audioQueue;
nsRefPtr<AudioDecodeRendezvous> barrier(new AudioDecodeRendezvous());
mDecoderReader->SetCallback(barrier);
while (1) {
mDecoderReader->RequestAudioData();
nsRefPtr<AudioData> audio;
if (NS_FAILED(barrier->Await(audio))) {
ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
return;
}
if (!audio) {
// End of stream.
break;
}
audioQueue.Push(audio);
}
mDecoderReader->Shutdown();
mDecoderReader->BreakCycles();
uint32_t frameCount = audioQueue.FrameCount();
uint32_t channelCount = mediaInfo.mAudio.mChannels;
uint32_t sampleRate = mediaInfo.mAudio.mRate;
if (!frameCount || !channelCount || !sampleRate) {
ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
return;
}
const uint32_t destSampleRate = mDecodeJob.mContext->SampleRate();
AutoResampler resampler;
uint32_t resampledFrames = frameCount;
if (sampleRate != destSampleRate) {
resampledFrames = static_cast<uint32_t>(
static_cast<uint64_t>(destSampleRate) *
static_cast<uint64_t>(frameCount) /
static_cast<uint64_t>(sampleRate)
);
resampler = speex_resampler_init(channelCount,
sampleRate,
destSampleRate,
SPEEX_RESAMPLER_QUALITY_DEFAULT, nullptr);
speex_resampler_skip_zeros(resampler);
resampledFrames += speex_resampler_get_output_latency(resampler);
}
// Allocate the channel buffers. Note that if we end up resampling, we may
// write fewer bytes than mResampledFrames to the output buffer, in which
// case mWriteIndex will tell us how many valid samples we have.
static const fallible_t fallible = fallible_t();
bool memoryAllocationSuccess = true;
if (!mDecodeJob.mChannelBuffers.SetLength(channelCount)) {
memoryAllocationSuccess = false;
} else {
for (uint32_t i = 0; i < channelCount; ++i) {
mDecodeJob.mChannelBuffers[i] = new(fallible) float[resampledFrames];
if (!mDecodeJob.mChannelBuffers[i]) {
memoryAllocationSuccess = false;
break;
}
}
}
if (!memoryAllocationSuccess) {
ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
return;
}
nsRefPtr<AudioData> audioData;
while ((audioData = audioQueue.PopFront())) {
audioData->EnsureAudioBuffer(); // could lead to a copy :(
AudioDataValue* bufferData = static_cast<AudioDataValue*>
(audioData->mAudioBuffer->Data());
if (sampleRate != destSampleRate) {
//.........这里部分代码省略.........