C++ AudioContext::SampleRate方法代码示例

      // Sets up |output| iff buffers are set in event handlers.
      void DispatchAudioProcessEvent(ScriptProcessorNode* aNode,
                                     AudioChunk* aOutput)
      {
        AudioContext* context = aNode->Context();
        if (!context) {
          return;
        }

        AutoJSAPI jsapi;
        if (NS_WARN_IF(!jsapi.Init(aNode->GetOwner()))) {
          return;
        }
        JSContext* cx = jsapi.cx();
        uint32_t inputChannelCount = aNode->ChannelCount();

        // Create the input buffer
        RefPtr<AudioBuffer> inputBuffer;
        if (mInputBuffer) {
          ErrorResult rv;
          inputBuffer =
            AudioBuffer::Create(context->GetOwner(), inputChannelCount,
                                aNode->BufferSize(), context->SampleRate(),
                                mInputBuffer.forget(), rv);
          if (rv.Failed()) {
            rv.SuppressException();
            return;
          }
        }

        // Ask content to produce data in the output buffer
        // Note that we always avoid creating the output buffer here, and we try to
        // avoid creating the input buffer as well.  The AudioProcessingEvent class
        // knows how to lazily create them if needed once the script tries to access
        // them.  Otherwise, we may be able to get away without creating them!
        RefPtr<AudioProcessingEvent> event =
          new AudioProcessingEvent(aNode, nullptr, nullptr);
        event->InitEvent(inputBuffer, inputChannelCount, mPlaybackTime);
        aNode->DispatchTrustedEvent(event);

        // Steal the output buffers if they have been set.
        // Don't create a buffer if it hasn't been used to return output;
        // FinishProducingOutputBuffer() will optimize output = null.
        // GetThreadSharedChannelsForRate() may also return null after OOM.
        if (event->HasOutputBuffer()) {
          ErrorResult rv;
          AudioBuffer* buffer = event->GetOutputBuffer(rv);
          // HasOutputBuffer() returning true means that GetOutputBuffer()
          // will not fail.
          MOZ_ASSERT(!rv.Failed());
          *aOutput = buffer->GetThreadSharedChannelsForRate(cx);
          MOZ_ASSERT(aOutput->IsNull() ||
                     aOutput->mBufferFormat == AUDIO_FORMAT_FLOAT32,
                     "AudioBuffers initialized from JS have float data");
        }
      }

/* static */
already_AddRefed<AudioNodeExternalInputStream>
AudioNodeExternalInputStream::Create(MediaStreamGraph* aGraph,
                                     AudioNodeEngine* aEngine) {
  AudioContext* ctx = aEngine->NodeMainThread()->Context();
  MOZ_ASSERT(NS_IsMainThread());
  MOZ_ASSERT(aGraph->GraphRate() == ctx->SampleRate());

  RefPtr<AudioNodeExternalInputStream> stream =
      new AudioNodeExternalInputStream(aEngine, aGraph->GraphRate());
  stream->mSuspendedCount += ctx->ShouldSuspendNewStream();
  aGraph->AddStream(stream);
  return stream.forget();
}

/* static */ already_AddRefed<AudioBuffer>
AudioBuffer::Constructor(const GlobalObject& aGlobal,
                         AudioContext& aAudioContext,
                         const AudioBufferOptions& aOptions,
                         ErrorResult& aRv)
{
  if (!aOptions.mNumberOfChannels) {
    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
    return nullptr;
  }

  float sampleRate = aOptions.mSampleRate.WasPassed()
                       ? aOptions.mSampleRate.Value()
                       : aAudioContext.SampleRate();
  return Create(&aAudioContext, aOptions.mNumberOfChannels, aOptions.mLength,
                sampleRate, aRv);
}