1 files changed, 215 insertions, 0 deletions
diff --git a/dom/media/AudioSegment.cpp b/dom/media/AudioSegment.cpp
new file mode 100644
index 000000000..b906f3462
--- /dev/null
+++ b/dom/media/AudioSegment.cpp
@@ -0,0 +1,215 @@
+/* -*- 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 "AudioSegment.h"
+
+#include "AudioMixer.h"
+#include "AudioChannelFormat.h"
+#include "Latency.h"
+#include <speex/speex_resampler.h>
+
+namespace mozilla {
+
+const uint8_t SilentChannel::gZeroChannel[MAX_AUDIO_SAMPLE_SIZE*SilentChannel::AUDIO_PROCESSING_FRAMES] = {0};
+
+template<>
+const float* SilentChannel::ZeroChannel<float>()
+{
+  return reinterpret_cast<const float*>(SilentChannel::gZeroChannel);
+}
+
+template<>
+const int16_t* SilentChannel::ZeroChannel<int16_t>()
+{
+  return reinterpret_cast<const int16_t*>(SilentChannel::gZeroChannel);
+}
+
+void
+AudioSegment::ApplyVolume(float aVolume)
+{
+  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
+    ci->mVolume *= aVolume;
+  }
+}
+
+void AudioSegment::ResampleChunks(SpeexResamplerState* aResampler, uint32_t aInRate, uint32_t aOutRate)
+{
+  if (mChunks.IsEmpty()) {
+    return;
+  }
+
+  MOZ_ASSERT(aResampler || IsNull(), "We can only be here without a resampler if this segment is null.");
+
+  AudioSampleFormat format = AUDIO_FORMAT_SILENCE;
+  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
+    if (ci->mBufferFormat != AUDIO_FORMAT_SILENCE) {
+      format = ci->mBufferFormat;
+    }
+  }
+
+  switch (format) {
+    // If the format is silence at this point, all the chunks are silent. The
+    // actual function we use does not matter, it's just a matter of changing
+    // the chunks duration.
+    case AUDIO_FORMAT_SILENCE:
+    case AUDIO_FORMAT_FLOAT32:
+      Resample<float>(aResampler, aInRate, aOutRate);
+    break;
+    case AUDIO_FORMAT_S16:
+      Resample<int16_t>(aResampler, aInRate, aOutRate);
+    break;
+    default:
+      MOZ_ASSERT(false);
+    break;
+  }
+}
+
+// This helps to to safely get a pointer to the position we want to start
+// writing a planar audio buffer, depending on the channel and the offset in the
+// buffer.
+static AudioDataValue*
+PointerForOffsetInChannel(AudioDataValue* aData, size_t aLengthSamples,
+                          uint32_t aChannelCount, uint32_t aChannel,
+                          uint32_t aOffsetSamples)
+{
+  size_t samplesPerChannel = aLengthSamples / aChannelCount;
+  size_t beginningOfChannel = samplesPerChannel * aChannel;
+  MOZ_ASSERT(aChannel * samplesPerChannel + aOffsetSamples < aLengthSamples,
+             "Offset request out of bounds.");
+  return aData + beginningOfChannel + aOffsetSamples;
+}
+
+void
+AudioSegment::Mix(AudioMixer& aMixer, uint32_t aOutputChannels,
+                  uint32_t aSampleRate)
+{
+  AutoTArray<AudioDataValue, SilentChannel::AUDIO_PROCESSING_FRAMES* GUESS_AUDIO_CHANNELS>
+  buf;
+  AutoTArray<const AudioDataValue*, GUESS_AUDIO_CHANNELS> channelData;
+  uint32_t offsetSamples = 0;
+  uint32_t duration = GetDuration();
+
+  if (duration <= 0) {
+    MOZ_ASSERT(duration == 0);
+    return;
+  }
+
+  uint32_t outBufferLength = duration * aOutputChannels;
+  buf.SetLength(outBufferLength);
+
+  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
+    AudioChunk& c = *ci;
+    uint32_t frames = c.mDuration;
+
+    // If the chunk is silent, simply write the right number of silence in the
+    // buffers.
+    if (c.mBufferFormat == AUDIO_FORMAT_SILENCE) {
+      for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
+        AudioDataValue* ptr =
+          PointerForOffsetInChannel(buf.Elements(), outBufferLength,
+                                    aOutputChannels, channel, offsetSamples);
+        PodZero(ptr, frames);
+      }
+    } else {
+      // Othewise, we need to upmix or downmix appropriately, depending on the
+      // desired input and output channels.
+      channelData.SetLength(c.mChannelData.Length());
+      for (uint32_t i = 0; i < channelData.Length(); ++i) {
+        channelData[i] = static_cast<const AudioDataValue*>(c.mChannelData[i]);
+      }
+      if (channelData.Length() < aOutputChannels) {
+        // Up-mix.
+        AudioChannelsUpMix(&channelData, aOutputChannels, SilentChannel::ZeroChannel<AudioDataValue>());
+        for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
+          AudioDataValue* ptr =
+            PointerForOffsetInChannel(buf.Elements(), outBufferLength,
+                                      aOutputChannels, channel, offsetSamples);
+          PodCopy(ptr, reinterpret_cast<const AudioDataValue*>(channelData[channel]),
+                  frames);
+        }
+        MOZ_ASSERT(channelData.Length() == aOutputChannels);
+      } else if (channelData.Length() > aOutputChannels) {
+        // Down mix.
+        AutoTArray<AudioDataValue*, GUESS_AUDIO_CHANNELS> outChannelPtrs;
+        outChannelPtrs.SetLength(aOutputChannels);
+        uint32_t offsetSamples = 0;
+        for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
+          outChannelPtrs[channel] =
+            PointerForOffsetInChannel(buf.Elements(), outBufferLength,
+                                      aOutputChannels, channel, offsetSamples);
+        }
+        AudioChannelsDownMix(channelData, outChannelPtrs.Elements(),
+                             aOutputChannels, frames);
+      } else {
+        // The channel count is already what we want, just copy it over.
+        for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
+          AudioDataValue* ptr =
+            PointerForOffsetInChannel(buf.Elements(), outBufferLength,
+                                      aOutputChannels, channel, offsetSamples);
+          PodCopy(ptr, reinterpret_cast<const AudioDataValue*>(channelData[channel]),
+                  frames);
+        }
+      }
+    }
+    offsetSamples += frames;
+  }
+
+  if (offsetSamples) {
+    MOZ_ASSERT(offsetSamples == outBufferLength / aOutputChannels,
+               "We forgot to write some samples?");
+    aMixer.Mix(buf.Elements(), aOutputChannels, offsetSamples, aSampleRate);
+  }
+}
+
+void
+AudioSegment::WriteTo(uint64_t aID, AudioMixer& aMixer, uint32_t aOutputChannels, uint32_t aSampleRate)
+{
+  AutoTArray<AudioDataValue,SilentChannel::AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> buf;
+  // Offset in the buffer that will be written to the mixer, in samples.
+  uint32_t offset = 0;
+
+  if (GetDuration() <= 0) {
+    MOZ_ASSERT(GetDuration() == 0);
+    return;
+  }
+
+  uint32_t outBufferLength = GetDuration() * aOutputChannels;
+  buf.SetLength(outBufferLength);
+
+
+  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
+    AudioChunk& c = *ci;
+
+    switch (c.mBufferFormat) {
+      case AUDIO_FORMAT_S16:
+        WriteChunk<int16_t>(c, aOutputChannels, buf.Elements() + offset);
+        break;
+      case AUDIO_FORMAT_FLOAT32:
+        WriteChunk<float>(c, aOutputChannels, buf.Elements() + offset);
+        break;
+      case AUDIO_FORMAT_SILENCE:
+        // The mixer is expecting interleaved data, so this is ok.
+        PodZero(buf.Elements() + offset, c.mDuration * aOutputChannels);
+        break;
+      default:
+        MOZ_ASSERT(false, "Not handled");
+    }
+
+    offset += c.mDuration * aOutputChannels;
+
+    if (!c.mTimeStamp.IsNull()) {
+      TimeStamp now = TimeStamp::Now();
+      // would be more efficient to c.mTimeStamp to ms on create time then pass here
+      LogTime(AsyncLatencyLogger::AudioMediaStreamTrack, aID,
+              (now - c.mTimeStamp).ToMilliseconds(), c.mTimeStamp);
+    }
+  }
+
+  if (offset) {
+    aMixer.Mix(buf.Elements(), aOutputChannels, offset / aOutputChannels, aSampleRate);
+  }
+}
+
+} // namespace mozilla