Add m-esr52 at 52.6.0

1 files changed, 462 insertions, 0 deletions

diff --git a/dom/media/encoder/OpusTrackEncoder.cpp b/dom/media/encoder/OpusTrackEncoder.cpp
new file mode 100644
index 000000000..c65d57788
--- /dev/null
+++ b/dom/media/encoder/OpusTrackEncoder.cpp
@@ -0,0 +1,462 @@
+/* -*- 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 "OpusTrackEncoder.h"
+#include "nsString.h"
+#include "GeckoProfiler.h"
+
+#include <opus/opus.h>
+
+#undef LOG
+#ifdef MOZ_WIDGET_GONK
+#include <android/log.h>
+#define LOG(args...) __android_log_print(ANDROID_LOG_INFO, "MediaEncoder", ## args);
+#else
+#define LOG(args, ...)
+#endif
+
+namespace mozilla {
+
+// The Opus format supports up to 8 channels, and supports multitrack audio up
+// to 255 channels, but the current implementation supports only mono and
+// stereo, and downmixes any more than that.
+static const int MAX_SUPPORTED_AUDIO_CHANNELS = 8;
+
+// http://www.opus-codec.org/docs/html_api-1.0.2/group__opus__encoder.html
+// In section "opus_encoder_init", channels must be 1 or 2 of input signal.
+static const int MAX_CHANNELS = 2;
+
+// A maximum data bytes for Opus to encode.
+static const int MAX_DATA_BYTES = 4096;
+
+// http://tools.ietf.org/html/draft-ietf-codec-oggopus-00#section-4
+// Second paragraph, " The granule position of an audio data page is in units
+// of PCM audio samples at a fixed rate of 48 kHz."
+static const int kOpusSamplingRate = 48000;
+
+// The duration of an Opus frame, and it must be 2.5, 5, 10, 20, 40 or 60 ms.
+static const int kFrameDurationMs  = 20;
+
+// The supported sampling rate of input signal (Hz),
+// must be one of the following. Will resampled to 48kHz otherwise.
+static const int kOpusSupportedInputSamplingRates[] =
+                   {8000, 12000, 16000, 24000, 48000};
+
+namespace {
+
+// An endian-neutral serialization of integers. Serializing T in little endian
+// format to aOutput, where T is a 16 bits or 32 bits integer.
+template<typename T>
+static void
+SerializeToBuffer(T aValue, nsTArray<uint8_t>* aOutput)
+{
+  for (uint32_t i = 0; i < sizeof(T); i++) {
+    aOutput->AppendElement((uint8_t)(0x000000ff & (aValue >> (i * 8))));
+  }
+}
+
+static inline void
+SerializeToBuffer(const nsCString& aComment, nsTArray<uint8_t>* aOutput)
+{
+  // Format of serializing a string to buffer is, the length of string (32 bits,
+  // little endian), and the string.
+  SerializeToBuffer((uint32_t)(aComment.Length()), aOutput);
+  aOutput->AppendElements(aComment.get(), aComment.Length());
+}
+
+
+static void
+SerializeOpusIdHeader(uint8_t aChannelCount, uint16_t aPreskip,
+                      uint32_t aInputSampleRate, nsTArray<uint8_t>* aOutput)
+{
+  // The magic signature, null terminator has to be stripped off from strings.
+  static const uint8_t magic[] = "OpusHead";
+  aOutput->AppendElements(magic, sizeof(magic) - 1);
+
+  // The version must always be 1 (8 bits, unsigned).
+  aOutput->AppendElement(1);
+
+  // Number of output channels (8 bits, unsigned).
+  aOutput->AppendElement(aChannelCount);
+
+  // Number of samples (at 48 kHz) to discard from the decoder output when
+  // starting playback (16 bits, unsigned, little endian).
+  SerializeToBuffer(aPreskip, aOutput);
+
+  // The sampling rate of input source (32 bits, unsigned, little endian).
+  SerializeToBuffer(aInputSampleRate, aOutput);
+
+  // Output gain, an encoder should set this field to zero (16 bits, signed,
+  // little endian).
+  SerializeToBuffer((int16_t)0, aOutput);
+
+  // Channel mapping family. Family 0 allows only 1 or 2 channels (8 bits,
+  // unsigned).
+  aOutput->AppendElement(0);
+}
+
+static void
+SerializeOpusCommentHeader(const nsCString& aVendor,
+                           const nsTArray<nsCString>& aComments,
+                           nsTArray<uint8_t>* aOutput)
+{
+  // The magic signature, null terminator has to be stripped off.
+  static const uint8_t magic[] = "OpusTags";
+  aOutput->AppendElements(magic, sizeof(magic) - 1);
+
+  // The vendor; Should append in the following order:
+  // vendor string length (32 bits, unsigned, little endian)
+  // vendor string.
+  SerializeToBuffer(aVendor, aOutput);
+
+  // Add comments; Should append in the following order:
+  // comment list length (32 bits, unsigned, little endian)
+  // comment #0 string length (32 bits, unsigned, little endian)
+  // comment #0 string
+  // comment #1 string length (32 bits, unsigned, little endian)
+  // comment #1 string ...
+  SerializeToBuffer((uint32_t)aComments.Length(), aOutput);
+  for (uint32_t i = 0; i < aComments.Length(); ++i) {
+    SerializeToBuffer(aComments[i], aOutput);
+  }
+}
+
+}  // Anonymous namespace.
+
+OpusTrackEncoder::OpusTrackEncoder()
+  : AudioTrackEncoder()
+  , mEncoder(nullptr)
+  , mLookahead(0)
+  , mResampler(nullptr)
+  , mOutputTimeStamp(0)
+{
+}
+
+OpusTrackEncoder::~OpusTrackEncoder()
+{
+  if (mEncoder) {
+    opus_encoder_destroy(mEncoder);
+  }
+  if (mResampler) {
+    speex_resampler_destroy(mResampler);
+    mResampler = nullptr;
+  }
+}
+
+nsresult
+OpusTrackEncoder::Init(int aChannels, int aSamplingRate)
+{
+  // This monitor is used to wake up other methods that are waiting for encoder
+  // to be completely initialized.
+  ReentrantMonitorAutoEnter mon(mReentrantMonitor);
+
+  NS_ENSURE_TRUE((aChannels <= MAX_SUPPORTED_AUDIO_CHANNELS) && (aChannels > 0),
+                 NS_ERROR_FAILURE);
+
+  // This version of encoder API only support 1 or 2 channels,
+  // So set the mChannels less or equal 2 and
+  // let InterleaveTrackData downmix pcm data.
+  mChannels = aChannels > MAX_CHANNELS ? MAX_CHANNELS : aChannels;
+
+  // Reject non-audio sample rates.
+  NS_ENSURE_TRUE(aSamplingRate >= 8000, NS_ERROR_INVALID_ARG);
+  NS_ENSURE_TRUE(aSamplingRate <= 192000, NS_ERROR_INVALID_ARG);
+
+  // According to www.opus-codec.org, creating an opus encoder requires the
+  // sampling rate of source signal be one of 8000, 12000, 16000, 24000, or
+  // 48000. If this constraint is not satisfied, we resample the input to 48kHz.
+  nsTArray<int> supportedSamplingRates;
+  supportedSamplingRates.AppendElements(kOpusSupportedInputSamplingRates,
+                         ArrayLength(kOpusSupportedInputSamplingRates));
+  if (!supportedSamplingRates.Contains(aSamplingRate)) {
+    int error;
+    mResampler = speex_resampler_init(mChannels,
+                                      aSamplingRate,
+                                      kOpusSamplingRate,
+                                      SPEEX_RESAMPLER_QUALITY_DEFAULT,
+                                      &error);
+
+    if (error != RESAMPLER_ERR_SUCCESS) {
+      return NS_ERROR_FAILURE;
+    }
+  }
+  mSamplingRate = aSamplingRate;
+  NS_ENSURE_TRUE(mSamplingRate > 0, NS_ERROR_FAILURE);
+
+  int error = 0;
+  mEncoder = opus_encoder_create(GetOutputSampleRate(), mChannels,
+                                 OPUS_APPLICATION_AUDIO, &error);
+
+
+  mInitialized = (error == OPUS_OK);
+
+  if (mAudioBitrate) {
+    opus_encoder_ctl(mEncoder, OPUS_SET_BITRATE(static_cast<int>(mAudioBitrate)));
+  }
+
+  mReentrantMonitor.NotifyAll();
+
+  return error == OPUS_OK ? NS_OK : NS_ERROR_FAILURE;
+}
+
+int
+OpusTrackEncoder::GetOutputSampleRate()
+{
+  return mResampler ? kOpusSamplingRate : mSamplingRate;
+}
+
+int
+OpusTrackEncoder::GetPacketDuration()
+{
+  return GetOutputSampleRate() * kFrameDurationMs / 1000;
+}
+
+already_AddRefed<TrackMetadataBase>
+OpusTrackEncoder::GetMetadata()
+{
+  PROFILER_LABEL("OpusTrackEncoder", "GetMetadata",
+    js::ProfileEntry::Category::OTHER);
+  {
+    // Wait if mEncoder is not initialized.
+    ReentrantMonitorAutoEnter mon(mReentrantMonitor);
+    while (!mCanceled && !mInitialized) {
+      mReentrantMonitor.Wait();
+    }
+  }
+
+  if (mCanceled || mEncodingComplete) {
+    return nullptr;
+  }
+
+  RefPtr<OpusMetadata> meta = new OpusMetadata();
+  meta->mChannels = mChannels;
+  meta->mSamplingFrequency = mSamplingRate;
+
+  mLookahead = 0;
+  int error = opus_encoder_ctl(mEncoder, OPUS_GET_LOOKAHEAD(&mLookahead));
+  if (error != OPUS_OK) {
+    mLookahead = 0;
+  }
+
+  // The ogg time stamping and pre-skip is always timed at 48000.
+  SerializeOpusIdHeader(mChannels, mLookahead * (kOpusSamplingRate /
+                        GetOutputSampleRate()), mSamplingRate,
+                        &meta->mIdHeader);
+
+  nsCString vendor;
+  vendor.AppendASCII(opus_get_version_string());
+
+  nsTArray<nsCString> comments;
+  comments.AppendElement(NS_LITERAL_CSTRING("ENCODER=Mozilla" MOZ_APP_UA_VERSION));
+
+  SerializeOpusCommentHeader(vendor, comments,
+                             &meta->mCommentHeader);
+
+  return meta.forget();
+}
+
+nsresult
+OpusTrackEncoder::GetEncodedTrack(EncodedFrameContainer& aData)
+{
+  PROFILER_LABEL("OpusTrackEncoder", "GetEncodedTrack",
+    js::ProfileEntry::Category::OTHER);
+  {
+    ReentrantMonitorAutoEnter mon(mReentrantMonitor);
+    // Wait until initialized or cancelled.
+    while (!mCanceled && !mInitialized) {
+      mReentrantMonitor.Wait();
+    }
+    if (mCanceled || mEncodingComplete) {
+      return NS_ERROR_FAILURE;
+    }
+  }
+
+  // calculation below depends on the truth that mInitialized is true.
+  MOZ_ASSERT(mInitialized);
+
+  bool wait = true;
+  int result = 0;
+  // Only wait once, then loop until we run out of packets of input data
+  while (result >= 0 && !mEncodingComplete) {
+    // re-sampled frames left last time which didn't fit into an Opus packet duration.
+    const int framesLeft = mResampledLeftover.Length() / mChannels;
+    // When framesLeft is 0, (GetPacketDuration() - framesLeft) is a multiple
+    // of kOpusSamplingRate. There is not precision loss in the integer division
+    // in computing framesToFetch. If frameLeft > 0, we need to add 1 to
+    // framesToFetch to ensure there will be at least n frames after re-sampling.
+    const int frameRoundUp = framesLeft ? 1 : 0;
+
+    MOZ_ASSERT(GetPacketDuration() >= framesLeft);
+    // Try to fetch m frames such that there will be n frames
+    // where (n + frameLeft) >= GetPacketDuration() after re-sampling.
+    const int framesToFetch = !mResampler ? GetPacketDuration()
+                              : (GetPacketDuration() - framesLeft) * mSamplingRate / kOpusSamplingRate
+                              + frameRoundUp;
+    {
+      // Move all the samples from mRawSegment to mSourceSegment. We only hold
+      // the monitor in this block.
+      ReentrantMonitorAutoEnter mon(mReentrantMonitor);
+
+      // Wait until enough raw data, end of stream or cancelled.
+      while (!mCanceled && mRawSegment.GetDuration() +
+             mSourceSegment.GetDuration() < framesToFetch &&
+             !mEndOfStream) {
+        if (wait) {
+          mReentrantMonitor.Wait();
+          wait = false;
+        } else {
+          goto done; // nested while's...
+        }
+      }
+
+      if (mCanceled) {
+        return NS_ERROR_FAILURE;
+      }
+
+      mSourceSegment.AppendFrom(&mRawSegment);
+
+      // Pad |mLookahead| samples to the end of source stream to prevent lost of
+      // original data, the pcm duration will be calculated at rate 48K later.
+      if (mEndOfStream && !mEosSetInEncoder) {
+        mEosSetInEncoder = true;
+        mSourceSegment.AppendNullData(mLookahead);
+      }
+    }
+
+    // Start encoding data.
+    AutoTArray<AudioDataValue, 9600> pcm;
+    pcm.SetLength(GetPacketDuration() * mChannels);
+    AudioSegment::ChunkIterator iter(mSourceSegment);
+    int frameCopied = 0;
+
+    while (!iter.IsEnded() && frameCopied < framesToFetch) {
+      AudioChunk chunk = *iter;
+
+      // Chunk to the required frame size.
+      int frameToCopy = chunk.GetDuration();
+      if (frameCopied + frameToCopy > framesToFetch) {
+        frameToCopy = framesToFetch - frameCopied;
+      }
+
+      if (!chunk.IsNull()) {
+        // Append the interleaved data to the end of pcm buffer.
+        AudioTrackEncoder::InterleaveTrackData(chunk, frameToCopy, mChannels,
+                                               pcm.Elements() + frameCopied * mChannels);
+      } else {
+        memset(pcm.Elements() + frameCopied * mChannels, 0,
+               frameToCopy * mChannels * sizeof(AudioDataValue));
+      }
+
+      frameCopied += frameToCopy;
+      iter.Next();
+    }
+
+    RefPtr<EncodedFrame> audiodata = new EncodedFrame();
+    audiodata->SetFrameType(EncodedFrame::OPUS_AUDIO_FRAME);
+    int framesInPCM = frameCopied;
+    if (mResampler) {
+      AutoTArray<AudioDataValue, 9600> resamplingDest;
+      // We want to consume all the input data, so we slightly oversize the
+      // resampled data buffer so we can fit the output data in. We cannot really
+      // predict the output frame count at each call.
+      uint32_t outframes = frameCopied * kOpusSamplingRate / mSamplingRate + 1;
+      uint32_t inframes = frameCopied;
+
+      resamplingDest.SetLength(outframes * mChannels);
+
+#if MOZ_SAMPLE_TYPE_S16
+      short* in = reinterpret_cast<short*>(pcm.Elements());
+      short* out = reinterpret_cast<short*>(resamplingDest.Elements());
+      speex_resampler_process_interleaved_int(mResampler, in, &inframes,
+                                              out, &outframes);
+#else
+      float* in = reinterpret_cast<float*>(pcm.Elements());
+      float* out = reinterpret_cast<float*>(resamplingDest.Elements());
+      speex_resampler_process_interleaved_float(mResampler, in, &inframes,
+                                                out, &outframes);
+#endif
+
+      MOZ_ASSERT(pcm.Length() >= mResampledLeftover.Length());
+      PodCopy(pcm.Elements(), mResampledLeftover.Elements(),
+              mResampledLeftover.Length());
+
+      uint32_t outframesToCopy = std::min(outframes,
+                                          static_cast<uint32_t>(GetPacketDuration() - framesLeft));
+
+      MOZ_ASSERT(pcm.Length() - mResampledLeftover.Length() >=
+                 outframesToCopy * mChannels);
+      PodCopy(pcm.Elements() + mResampledLeftover.Length(),
+              resamplingDest.Elements(), outframesToCopy * mChannels);
+      int frameLeftover = outframes - outframesToCopy;
+      mResampledLeftover.SetLength(frameLeftover * mChannels);
+      PodCopy(mResampledLeftover.Elements(),
+              resamplingDest.Elements() + outframesToCopy * mChannels,
+              mResampledLeftover.Length());
+      // This is always at 48000Hz.
+      framesInPCM = framesLeft + outframesToCopy;
+      audiodata->SetDuration(framesInPCM);
+    } else {
+      // The ogg time stamping and pre-skip is always timed at 48000.
+      audiodata->SetDuration(frameCopied * (kOpusSamplingRate / mSamplingRate));
+    }
+
+    // Remove the raw data which has been pulled to pcm buffer.
+    // The value of frameCopied should equal to (or smaller than, if eos)
+    // GetPacketDuration().
+    mSourceSegment.RemoveLeading(frameCopied);
+
+    // Has reached the end of input stream and all queued data has pulled for
+    // encoding.
+    if (mSourceSegment.GetDuration() == 0 && mEosSetInEncoder) {
+      mEncodingComplete = true;
+      LOG("[Opus] Done encoding.");
+    }
+
+    MOZ_ASSERT(mEosSetInEncoder || framesInPCM == GetPacketDuration());
+
+    // Append null data to pcm buffer if the leftover data is not enough for
+    // opus encoder.
+    if (framesInPCM < GetPacketDuration() && mEosSetInEncoder) {
+      PodZero(pcm.Elements() + framesInPCM * mChannels,
+              (GetPacketDuration() - framesInPCM) * mChannels);
+    }
+    nsTArray<uint8_t> frameData;
+    // Encode the data with Opus Encoder.
+    frameData.SetLength(MAX_DATA_BYTES);
+    // result is returned as opus error code if it is negative.
+    result = 0;
+#ifdef MOZ_SAMPLE_TYPE_S16
+    const opus_int16* pcmBuf = static_cast<opus_int16*>(pcm.Elements());
+    result = opus_encode(mEncoder, pcmBuf, GetPacketDuration(),
+                         frameData.Elements(), MAX_DATA_BYTES);
+#else
+    const float* pcmBuf = static_cast<float*>(pcm.Elements());
+    result = opus_encode_float(mEncoder, pcmBuf, GetPacketDuration(),
+                               frameData.Elements(), MAX_DATA_BYTES);
+#endif
+    frameData.SetLength(result >= 0 ? result : 0);
+
+    if (result < 0) {
+      LOG("[Opus] Fail to encode data! Result: %s.", opus_strerror(result));
+    }
+    if (mEncodingComplete) {
+      if (mResampler) {
+        speex_resampler_destroy(mResampler);
+        mResampler = nullptr;
+      }
+      mResampledLeftover.SetLength(0);
+    }
+
+    audiodata->SwapInFrameData(frameData);
+    // timestamp should be the time of the first sample
+    audiodata->SetTimeStamp(mOutputTimeStamp);
+    mOutputTimeStamp += FramesToUsecs(GetPacketDuration(), kOpusSamplingRate).value();
+    LOG("[Opus] mOutputTimeStamp %lld.",mOutputTimeStamp);
+    aData.AppendEncodedFrame(audiodata);
+  }
+done:
+  return result >= 0 ? NS_OK : NS_ERROR_FAILURE;
+}
+
+} // namespace mozilla