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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /dom/media/encoder/OpusTrackEncoder.cpp
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'dom/media/encoder/OpusTrackEncoder.cpp')
-rw-r--r--dom/media/encoder/OpusTrackEncoder.cpp462
1 files changed, 462 insertions, 0 deletions
diff --git a/dom/media/encoder/OpusTrackEncoder.cpp b/dom/media/encoder/OpusTrackEncoder.cpp
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+++ b/dom/media/encoder/OpusTrackEncoder.cpp
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+/* -*- 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