Adding upstream version 115.8.0esr.upstream/115.8.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 454 insertions, 0 deletions

diff --git a/dom/media/encoder/OpusTrackEncoder.cpp b/dom/media/encoder/OpusTrackEncoder.cpp
new file mode 100644
index 0000000000..16b71d378e
--- /dev/null
+++ b/dom/media/encoder/OpusTrackEncoder.cpp
@@ -0,0 +1,454 @@
+/* -*- 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 "mozilla/CheckedInt.h"
+#include "mozilla/ProfilerLabels.h"
+#include "VideoUtils.h"
+
+#include <opus/opus.h>
+
+#define LOG(args, ...)
+
+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.
+constexpr 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.
+constexpr int MAX_CHANNELS = 2;
+
+// A maximum data bytes for Opus to encode.
+constexpr 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."
+constexpr int kOpusSamplingRate = 48000;
+
+// The duration of an Opus frame, and it must be 2.5, 5, 10, 20, 40 or 60 ms.
+constexpr int kFrameDurationMs = 20;
+
+// The supported sampling rate of input signal (Hz),
+// must be one of the following. Will resampled to 48kHz otherwise.
+constexpr 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.
+  constexpr 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.
+  constexpr 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);
+  }
+}
+
+bool IsSampleRateSupported(TrackRate aSampleRate) {
+  // 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.
+  AutoTArray<int, 5> supportedSamplingRates;
+  supportedSamplingRates.AppendElements(
+      kOpusSupportedInputSamplingRates,
+      ArrayLength(kOpusSupportedInputSamplingRates));
+  return supportedSamplingRates.Contains(aSampleRate);
+}
+
+}  // Anonymous namespace.
+
+OpusTrackEncoder::OpusTrackEncoder(TrackRate aTrackRate,
+                                   MediaQueue<EncodedFrame>& aEncodedDataQueue)
+    : AudioTrackEncoder(aTrackRate, aEncodedDataQueue),
+      mOutputSampleRate(IsSampleRateSupported(aTrackRate) ? aTrackRate
+                                                          : kOpusSamplingRate),
+      mEncoder(nullptr),
+      mLookahead(0),
+      mLookaheadWritten(0),
+      mResampler(nullptr),
+      mNumOutputFrames(0) {}
+
+OpusTrackEncoder::~OpusTrackEncoder() {
+  if (mEncoder) {
+    opus_encoder_destroy(mEncoder);
+  }
+  if (mResampler) {
+    speex_resampler_destroy(mResampler);
+    mResampler = nullptr;
+  }
+}
+
+nsresult OpusTrackEncoder::Init(int aChannels) {
+  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(mTrackRate >= 8000, NS_ERROR_INVALID_ARG);
+  NS_ENSURE_TRUE(mTrackRate <= 192000, NS_ERROR_INVALID_ARG);
+
+  if (NeedsResampler()) {
+    int error;
+    mResampler = speex_resampler_init(mChannels, mTrackRate, kOpusSamplingRate,
+                                      SPEEX_RESAMPLER_QUALITY_DEFAULT, &error);
+
+    if (error != RESAMPLER_ERR_SUCCESS) {
+      return NS_ERROR_FAILURE;
+    }
+  }
+
+  int error = 0;
+  mEncoder = opus_encoder_create(mOutputSampleRate, mChannels,
+                                 OPUS_APPLICATION_AUDIO, &error);
+
+  if (error != OPUS_OK) {
+    return NS_ERROR_FAILURE;
+  }
+
+  if (mAudioBitrate) {
+    int bps = static_cast<int>(
+        std::min<uint32_t>(mAudioBitrate, std::numeric_limits<int>::max()));
+    error = opus_encoder_ctl(mEncoder, OPUS_SET_BITRATE(bps));
+    if (error != OPUS_OK) {
+      return NS_ERROR_FAILURE;
+    }
+  }
+
+  // In the case of Opus we need to calculate the codec delay based on the
+  // pre-skip. For more information see:
+  // https://tools.ietf.org/html/rfc7845#section-4.2
+  error = opus_encoder_ctl(mEncoder, OPUS_GET_LOOKAHEAD(&mLookahead));
+  if (error != OPUS_OK) {
+    mLookahead = 0;
+    return NS_ERROR_FAILURE;
+  }
+
+  SetInitialized();
+
+  return NS_OK;
+}
+
+int OpusTrackEncoder::GetLookahead() const {
+  return mLookahead * kOpusSamplingRate / mOutputSampleRate;
+}
+
+int OpusTrackEncoder::NumInputFramesPerPacket() const {
+  return mTrackRate * kFrameDurationMs / 1000;
+}
+
+int OpusTrackEncoder::NumOutputFramesPerPacket() const {
+  return mOutputSampleRate * kFrameDurationMs / 1000;
+}
+
+bool OpusTrackEncoder::NeedsResampler() const {
+  // A resampler is needed when mTrackRate is not supported by the opus encoder.
+  // This is equivalent to !IsSampleRateSupported(mTrackRate) but less cycles.
+  return mTrackRate != mOutputSampleRate &&
+         mOutputSampleRate == kOpusSamplingRate;
+}
+
+already_AddRefed<TrackMetadataBase> OpusTrackEncoder::GetMetadata() {
+  AUTO_PROFILER_LABEL("OpusTrackEncoder::GetMetadata", OTHER);
+
+  MOZ_ASSERT(mInitialized);
+
+  if (!mInitialized) {
+    return nullptr;
+  }
+
+  RefPtr<OpusMetadata> meta = new OpusMetadata();
+  meta->mChannels = mChannels;
+  meta->mSamplingFrequency = mTrackRate;
+
+  // Ogg and Webm timestamps are always sampled at 48k for Opus.
+  SerializeOpusIdHeader(mChannels,
+                        mLookahead * (kOpusSamplingRate / mOutputSampleRate),
+                        mTrackRate, &meta->mIdHeader);
+
+  nsCString vendor;
+  vendor.AppendASCII(opus_get_version_string());
+
+  nsTArray<nsCString> comments;
+  comments.AppendElement(
+      nsLiteralCString("ENCODER=Mozilla" MOZ_APP_UA_VERSION));
+
+  SerializeOpusCommentHeader(vendor, comments, &meta->mCommentHeader);
+
+  return meta.forget();
+}
+
+nsresult OpusTrackEncoder::Encode(AudioSegment* aSegment) {
+  AUTO_PROFILER_LABEL("OpusTrackEncoder::Encode", OTHER);
+
+  MOZ_ASSERT(aSegment);
+  MOZ_ASSERT(mInitialized || mCanceled);
+
+  if (mCanceled || IsEncodingComplete()) {
+    return NS_ERROR_FAILURE;
+  }
+
+  if (!mInitialized) {
+    // calculation below depends on the truth that mInitialized is true.
+    return NS_ERROR_FAILURE;
+  }
+
+  int result = 0;
+  // Loop until we run out of packets of input data
+  while (result >= 0 && !IsEncodingComplete()) {
+    // re-sampled frames left last time which didn't fit into an Opus packet
+    // duration.
+    const int framesLeft = mResampledLeftover.Length() / mChannels;
+    MOZ_ASSERT(NumOutputFramesPerPacket() >= framesLeft);
+    // Fetch input frames such that there will be n frames where (n +
+    // framesLeft) >= NumOutputFramesPerPacket() after re-sampling.
+    const int framesToFetch = NumInputFramesPerPacket() -
+                              (framesLeft * mTrackRate / kOpusSamplingRate) +
+                              (NeedsResampler() ? 1 : 0);
+
+    if (!mEndOfStream && aSegment->GetDuration() < framesToFetch) {
+      // Not enough raw data
+      return NS_OK;
+    }
+
+    // Start encoding data.
+    AutoTArray<AudioDataValue, 9600> pcm;
+    pcm.SetLength(NumOutputFramesPerPacket() * mChannels);
+
+    int frameCopied = 0;
+
+    for (AudioSegment::ChunkIterator iter(*aSegment);
+         !iter.IsEnded() && frameCopied < framesToFetch; iter.Next()) {
+      AudioChunk chunk = *iter;
+
+      // Chunk to the required frame size.
+      TrackTime frameToCopy =
+          std::min(chunk.GetDuration(),
+                   static_cast<TrackTime>(framesToFetch - frameCopied));
+
+      // Possible greatest value of framesToFetch = 3844: see
+      // https://bugzilla.mozilla.org/show_bug.cgi?id=1349421#c8. frameToCopy
+      // should not be able to exceed this value.
+      MOZ_ASSERT(frameToCopy <= 3844, "frameToCopy exceeded expected range");
+
+      if (!chunk.IsNull()) {
+        // Append the interleaved data to the end of pcm buffer.
+        AudioTrackEncoder::InterleaveTrackData(
+            chunk, frameToCopy, mChannels,
+            pcm.Elements() + frameCopied * mChannels);
+      } else {
+        CheckedInt<int> memsetLength =
+            CheckedInt<int>(frameToCopy) * mChannels * sizeof(AudioDataValue);
+        if (!memsetLength.isValid()) {
+          // This should never happen, but we use a defensive check because
+          // we really don't want a bad memset
+          MOZ_ASSERT_UNREACHABLE("memsetLength invalid!");
+          return NS_ERROR_FAILURE;
+        }
+        memset(pcm.Elements() + frameCopied * mChannels, 0,
+               memsetLength.value());
+      }
+
+      frameCopied += frameToCopy;
+    }
+
+    // Possible greatest value of framesToFetch = 3844: see
+    // https://bugzilla.mozilla.org/show_bug.cgi?id=1349421#c8. frameCopied
+    // should not be able to exceed this value.
+    MOZ_ASSERT(frameCopied <= 3844, "frameCopied exceeded expected range");
+
+    int framesInPCM = frameCopied;
+    if (mResampler) {
+      AutoTArray<AudioDataValue, 9600> resamplingDest;
+      uint32_t inframes = frameCopied;
+      uint32_t outframes = inframes * kOpusSamplingRate / mTrackRate + 1;
+
+      // 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.
+      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>(NumOutputFramesPerPacket() - 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;
+    }
+
+    // Remove the raw data which has been pulled to pcm buffer.
+    // The value of frameCopied should be equal to (or smaller than, if eos)
+    // NumOutputFramesPerPacket().
+    aSegment->RemoveLeading(frameCopied);
+
+    // Has reached the end of input stream and all queued data has pulled for
+    // encoding.
+    bool isFinalPacket = false;
+    if (aSegment->GetDuration() == 0 && mEndOfStream &&
+        framesInPCM < NumOutputFramesPerPacket()) {
+      // Pad |mLookahead| samples to the end of the track to prevent loss of
+      // original data.
+      const int toWrite = std::min(mLookahead - mLookaheadWritten,
+                                   NumOutputFramesPerPacket() - framesInPCM);
+      PodZero(pcm.Elements() + framesInPCM * mChannels, toWrite * mChannels);
+      mLookaheadWritten += toWrite;
+      framesInPCM += toWrite;
+      if (mLookaheadWritten == mLookahead) {
+        isFinalPacket = true;
+      }
+    }
+
+    MOZ_ASSERT_IF(!isFinalPacket, framesInPCM == NumOutputFramesPerPacket());
+
+    // Append null data to pcm buffer if the leftover data is not enough for
+    // opus encoder.
+    if (framesInPCM < NumOutputFramesPerPacket() && isFinalPacket) {
+      PodZero(pcm.Elements() + framesInPCM * mChannels,
+              (NumOutputFramesPerPacket() - framesInPCM) * mChannels);
+    }
+    auto frameData = MakeRefPtr<EncodedFrame::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, NumOutputFramesPerPacket(),
+                         frameData->Elements(), MAX_DATA_BYTES);
+#else
+    const float* pcmBuf = static_cast<float*>(pcm.Elements());
+    result = opus_encode_float(mEncoder, pcmBuf, NumOutputFramesPerPacket(),
+                               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 (isFinalPacket) {
+      if (mResampler) {
+        speex_resampler_destroy(mResampler);
+        mResampler = nullptr;
+      }
+      mResampledLeftover.SetLength(0);
+    }
+
+    // timestamp should be the time of the first sample
+    mEncodedDataQueue.Push(MakeAndAddRef<EncodedFrame>(
+        media::TimeUnit(mNumOutputFrames + mLookahead, mOutputSampleRate),
+        static_cast<uint64_t>(framesInPCM) * kOpusSamplingRate /
+            mOutputSampleRate,
+        kOpusSamplingRate, EncodedFrame::OPUS_AUDIO_FRAME,
+        std::move(frameData)));
+
+    mNumOutputFrames += NumOutputFramesPerPacket();
+    LOG("[Opus] mOutputTimeStamp %.3f.",
+        media::TimeUnit(mNumOutputFrames, mOutputSampleRate).ToSeconds());
+
+    if (isFinalPacket) {
+      LOG("[Opus] Done encoding.");
+      mEncodedDataQueue.Finish();
+    }
+  }
+
+  return result >= 0 ? NS_OK : NS_ERROR_FAILURE;
+}
+
+}  // namespace mozilla
+
+#undef LOG