Add m-esr52 at 52.6.0

1 files changed, 238 insertions, 0 deletions

diff --git a/dom/media/webaudio/AudioNodeExternalInputStream.cpp b/dom/media/webaudio/AudioNodeExternalInputStream.cpp
new file mode 100644
index 000000000..2dff1488b
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeExternalInputStream.cpp
@@ -0,0 +1,238 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AlignedTArray.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeExternalInputStream.h"
+#include "AudioChannelFormat.h"
+#include "mozilla/dom/MediaStreamAudioSourceNode.h"
+
+using namespace mozilla::dom;
+
+namespace mozilla {
+
+AudioNodeExternalInputStream::AudioNodeExternalInputStream(AudioNodeEngine* aEngine, TrackRate aSampleRate)
+  : AudioNodeStream(aEngine, NO_STREAM_FLAGS, aSampleRate)
+{
+  MOZ_COUNT_CTOR(AudioNodeExternalInputStream);
+}
+
+AudioNodeExternalInputStream::~AudioNodeExternalInputStream()
+{
+  MOZ_COUNT_DTOR(AudioNodeExternalInputStream);
+}
+
+/* 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();
+}
+
+/**
+ * Copies the data in aInput to aOffsetInBlock within aBlock.
+ * aBlock must have been allocated with AllocateInputBlock and have a channel
+ * count that's a superset of the channels in aInput.
+ */
+template <typename T>
+static void
+CopyChunkToBlock(AudioChunk& aInput, AudioBlock *aBlock,
+                 uint32_t aOffsetInBlock)
+{
+  uint32_t blockChannels = aBlock->ChannelCount();
+  AutoTArray<const T*,2> channels;
+  if (aInput.IsNull()) {
+    channels.SetLength(blockChannels);
+    PodZero(channels.Elements(), blockChannels);
+  } else {
+    const nsTArray<const T*>& inputChannels = aInput.ChannelData<T>();
+    channels.SetLength(inputChannels.Length());
+    PodCopy(channels.Elements(), inputChannels.Elements(), channels.Length());
+    if (channels.Length() != blockChannels) {
+      // We only need to upmix here because aBlock's channel count has been
+      // chosen to be a superset of the channel count of every chunk.
+      AudioChannelsUpMix(&channels, blockChannels, static_cast<T*>(nullptr));
+    }
+  }
+
+  for (uint32_t c = 0; c < blockChannels; ++c) {
+    float* outputData = aBlock->ChannelFloatsForWrite(c) + aOffsetInBlock;
+    if (channels[c]) {
+      ConvertAudioSamplesWithScale(channels[c], outputData, aInput.GetDuration(), aInput.mVolume);
+    } else {
+      PodZero(outputData, aInput.GetDuration());
+    }
+  }
+}
+
+/**
+ * Converts the data in aSegment to a single chunk aBlock. aSegment must have
+ * duration WEBAUDIO_BLOCK_SIZE. aFallbackChannelCount is a superset of the
+ * channels in every chunk of aSegment. aBlock must be float format or null.
+ */
+static void ConvertSegmentToAudioBlock(AudioSegment* aSegment,
+                                       AudioBlock* aBlock,
+                                       int32_t aFallbackChannelCount)
+{
+  NS_ASSERTION(aSegment->GetDuration() == WEBAUDIO_BLOCK_SIZE, "Bad segment duration");
+
+  {
+    AudioSegment::ChunkIterator ci(*aSegment);
+    NS_ASSERTION(!ci.IsEnded(), "Should be at least one chunk!");
+    if (ci->GetDuration() == WEBAUDIO_BLOCK_SIZE &&
+        (ci->IsNull() || ci->mBufferFormat == AUDIO_FORMAT_FLOAT32)) {
+
+      bool aligned = true;
+      for (size_t i = 0; i < ci->mChannelData.Length(); ++i) {
+        if (!IS_ALIGNED16(ci->mChannelData[i])) {
+            aligned = false;
+            break;
+        }
+      }
+
+      // Return this chunk directly to avoid copying data.
+      if (aligned) {
+        *aBlock = *ci;
+        return;
+      }
+    }
+  }
+
+  aBlock->AllocateChannels(aFallbackChannelCount);
+
+  uint32_t duration = 0;
+  for (AudioSegment::ChunkIterator ci(*aSegment); !ci.IsEnded(); ci.Next()) {
+    switch (ci->mBufferFormat) {
+      case AUDIO_FORMAT_S16: {
+        CopyChunkToBlock<int16_t>(*ci, aBlock, duration);
+        break;
+      }
+      case AUDIO_FORMAT_FLOAT32: {
+        CopyChunkToBlock<float>(*ci, aBlock, duration);
+        break;
+      }
+      case AUDIO_FORMAT_SILENCE: {
+        // The actual type of the sample does not matter here, but we still need
+        // to send some audio to the graph.
+        CopyChunkToBlock<float>(*ci, aBlock, duration);
+        break;
+      }
+    }
+    duration += ci->GetDuration();
+  }
+}
+
+void
+AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
+                                           uint32_t aFlags)
+{
+  // According to spec, number of outputs is always 1.
+  MOZ_ASSERT(mLastChunks.Length() == 1);
+
+  // GC stuff can result in our input stream being destroyed before this stream.
+  // Handle that.
+  if (!IsEnabled() || mInputs.IsEmpty() || mPassThrough) {
+    mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  MOZ_ASSERT(mInputs.Length() == 1);
+
+  MediaStream* source = mInputs[0]->GetSource();
+  AutoTArray<AudioSegment,1> audioSegments;
+  uint32_t inputChannels = 0;
+  for (StreamTracks::TrackIter tracks(source->mTracks);
+       !tracks.IsEnded(); tracks.Next()) {
+    const StreamTracks::Track& inputTrack = *tracks;
+    if (!mInputs[0]->PassTrackThrough(tracks->GetID())) {
+      continue;
+    }
+
+    if (inputTrack.GetSegment()->GetType() == MediaSegment::VIDEO) {
+      MOZ_ASSERT(false, "AudioNodeExternalInputStream shouldn't have video tracks");
+      continue;
+    }
+
+    const AudioSegment& inputSegment =
+        *static_cast<AudioSegment*>(inputTrack.GetSegment());
+    if (inputSegment.IsNull()) {
+      continue;
+    }
+
+    AudioSegment& segment = *audioSegments.AppendElement();
+    GraphTime next;
+    for (GraphTime t = aFrom; t < aTo; t = next) {
+      MediaInputPort::InputInterval interval = mInputs[0]->GetNextInputInterval(t);
+      interval.mEnd = std::min(interval.mEnd, aTo);
+      if (interval.mStart >= interval.mEnd)
+        break;
+      next = interval.mEnd;
+
+      // We know this stream does not block during the processing interval ---
+      // we're not finished, we don't underrun, and we're not suspended.
+      StreamTime outputStart = GraphTimeToStreamTime(interval.mStart);
+      StreamTime outputEnd = GraphTimeToStreamTime(interval.mEnd);
+      StreamTime ticks = outputEnd - outputStart;
+
+      if (interval.mInputIsBlocked) {
+        segment.AppendNullData(ticks);
+      } else {
+        // The input stream is not blocked in this interval, so no need to call
+        // GraphTimeToStreamTimeWithBlocking.
+        StreamTime inputStart =
+          std::min(inputSegment.GetDuration(),
+                   source->GraphTimeToStreamTime(interval.mStart));
+        StreamTime inputEnd =
+          std::min(inputSegment.GetDuration(),
+                   source->GraphTimeToStreamTime(interval.mEnd));
+
+        segment.AppendSlice(inputSegment, inputStart, inputEnd);
+        // Pad if we're looking past the end of the track
+        segment.AppendNullData(ticks - (inputEnd - inputStart));
+      }
+    }
+
+    for (AudioSegment::ChunkIterator iter(segment); !iter.IsEnded(); iter.Next()) {
+      inputChannels = GetAudioChannelsSuperset(inputChannels, iter->ChannelCount());
+    }
+  }
+
+  uint32_t accumulateIndex = 0;
+  if (inputChannels) {
+    DownmixBufferType downmixBuffer;
+    ASSERT_ALIGNED16(downmixBuffer.Elements());
+    for (uint32_t i = 0; i < audioSegments.Length(); ++i) {
+      AudioBlock tmpChunk;
+      ConvertSegmentToAudioBlock(&audioSegments[i], &tmpChunk, inputChannels);
+      if (!tmpChunk.IsNull()) {
+        if (accumulateIndex == 0) {
+          mLastChunks[0].AllocateChannels(inputChannels);
+        }
+        AccumulateInputChunk(accumulateIndex, tmpChunk, &mLastChunks[0], &downmixBuffer);
+        accumulateIndex++;
+      }
+    }
+  }
+  if (accumulateIndex == 0) {
+    mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+}
+
+bool
+AudioNodeExternalInputStream::IsEnabled()
+{
+  return ((MediaStreamAudioSourceNodeEngine*)Engine())->IsEnabled();
+}
+
+} // namespace mozilla