460 lines
16 KiB
C++
460 lines
16 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "OpusTrackEncoder.h"
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#include "VideoUtils.h"
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#include "nsString.h"
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#include "GeckoProfiler.h"
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#include <opus/opus.h>
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#undef LOG
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#define LOG(args, ...)
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namespace mozilla {
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// The Opus format supports up to 8 channels, and supports multitrack audio up
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// to 255 channels, but the current implementation supports only mono and
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// stereo, and downmixes any more than that.
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static const int MAX_SUPPORTED_AUDIO_CHANNELS = 8;
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// http://www.opus-codec.org/docs/html_api-1.0.2/group__opus__encoder.html
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// In section "opus_encoder_init", channels must be 1 or 2 of input signal.
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static const int MAX_CHANNELS = 2;
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// A maximum data bytes for Opus to encode.
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static const int MAX_DATA_BYTES = 4096;
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// http://tools.ietf.org/html/draft-ietf-codec-oggopus-00#section-4
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// Second paragraph, " The granule position of an audio data page is in units
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// of PCM audio samples at a fixed rate of 48 kHz."
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static const int kOpusSamplingRate = 48000;
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// The duration of an Opus frame, and it must be 2.5, 5, 10, 20, 40 or 60 ms.
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static const int kFrameDurationMs = 20;
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// The supported sampling rate of input signal (Hz),
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// must be one of the following. Will resampled to 48kHz otherwise.
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static const int kOpusSupportedInputSamplingRates[] =
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{8000, 12000, 16000, 24000, 48000};
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namespace {
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// An endian-neutral serialization of integers. Serializing T in little endian
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// format to aOutput, where T is a 16 bits or 32 bits integer.
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template<typename T>
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static void
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SerializeToBuffer(T aValue, nsTArray<uint8_t>* aOutput)
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{
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for (uint32_t i = 0; i < sizeof(T); i++) {
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aOutput->AppendElement((uint8_t)(0x000000ff & (aValue >> (i * 8))));
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}
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}
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static inline void
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SerializeToBuffer(const nsCString& aComment, nsTArray<uint8_t>* aOutput)
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{
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// Format of serializing a string to buffer is, the length of string (32 bits,
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// little endian), and the string.
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SerializeToBuffer((uint32_t)(aComment.Length()), aOutput);
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aOutput->AppendElements(aComment.get(), aComment.Length());
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}
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static void
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SerializeOpusIdHeader(uint8_t aChannelCount, uint16_t aPreskip,
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uint32_t aInputSampleRate, nsTArray<uint8_t>* aOutput)
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{
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// The magic signature, null terminator has to be stripped off from strings.
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static const uint8_t magic[] = "OpusHead";
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aOutput->AppendElements(magic, sizeof(magic) - 1);
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// The version must always be 1 (8 bits, unsigned).
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aOutput->AppendElement(1);
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// Number of output channels (8 bits, unsigned).
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aOutput->AppendElement(aChannelCount);
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// Number of samples (at 48 kHz) to discard from the decoder output when
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// starting playback (16 bits, unsigned, little endian).
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SerializeToBuffer(aPreskip, aOutput);
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// The sampling rate of input source (32 bits, unsigned, little endian).
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SerializeToBuffer(aInputSampleRate, aOutput);
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// Output gain, an encoder should set this field to zero (16 bits, signed,
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// little endian).
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SerializeToBuffer((int16_t)0, aOutput);
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// Channel mapping family. Family 0 allows only 1 or 2 channels (8 bits,
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// unsigned).
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aOutput->AppendElement(0);
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}
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static void
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SerializeOpusCommentHeader(const nsCString& aVendor,
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const nsTArray<nsCString>& aComments,
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nsTArray<uint8_t>* aOutput)
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{
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// The magic signature, null terminator has to be stripped off.
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static const uint8_t magic[] = "OpusTags";
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aOutput->AppendElements(magic, sizeof(magic) - 1);
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// The vendor; Should append in the following order:
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// vendor string length (32 bits, unsigned, little endian)
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// vendor string.
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SerializeToBuffer(aVendor, aOutput);
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// Add comments; Should append in the following order:
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// comment list length (32 bits, unsigned, little endian)
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// comment #0 string length (32 bits, unsigned, little endian)
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// comment #0 string
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// comment #1 string length (32 bits, unsigned, little endian)
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// comment #1 string ...
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SerializeToBuffer((uint32_t)aComments.Length(), aOutput);
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for (uint32_t i = 0; i < aComments.Length(); ++i) {
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SerializeToBuffer(aComments[i], aOutput);
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}
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}
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} // Anonymous namespace.
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OpusTrackEncoder::OpusTrackEncoder()
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: AudioTrackEncoder()
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, mEncoder(nullptr)
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, mLookahead(0)
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, mResampler(nullptr)
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, mOutputTimeStamp(0)
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{
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}
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OpusTrackEncoder::~OpusTrackEncoder()
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{
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if (mEncoder) {
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opus_encoder_destroy(mEncoder);
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}
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if (mResampler) {
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speex_resampler_destroy(mResampler);
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mResampler = nullptr;
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}
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}
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nsresult
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OpusTrackEncoder::Init(int aChannels, int aSamplingRate)
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{
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// This monitor is used to wake up other methods that are waiting for encoder
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// to be completely initialized.
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ReentrantMonitorAutoEnter mon(mReentrantMonitor);
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NS_ENSURE_TRUE((aChannels <= MAX_SUPPORTED_AUDIO_CHANNELS) && (aChannels > 0),
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NS_ERROR_FAILURE);
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// This version of encoder API only support 1 or 2 channels,
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// So set the mChannels less or equal 2 and
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// let InterleaveTrackData downmix pcm data.
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mChannels = aChannels > MAX_CHANNELS ? MAX_CHANNELS : aChannels;
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// Reject non-audio sample rates.
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NS_ENSURE_TRUE(aSamplingRate >= 8000, NS_ERROR_INVALID_ARG);
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NS_ENSURE_TRUE(aSamplingRate <= 192000, NS_ERROR_INVALID_ARG);
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// According to www.opus-codec.org, creating an opus encoder requires the
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// sampling rate of source signal be one of 8000, 12000, 16000, 24000, or
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// 48000. If this constraint is not satisfied, we resample the input to 48kHz.
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nsTArray<int> supportedSamplingRates;
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supportedSamplingRates.AppendElements(kOpusSupportedInputSamplingRates,
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ArrayLength(kOpusSupportedInputSamplingRates));
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if (!supportedSamplingRates.Contains(aSamplingRate)) {
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int error;
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mResampler = speex_resampler_init(mChannels,
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aSamplingRate,
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kOpusSamplingRate,
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SPEEX_RESAMPLER_QUALITY_DEFAULT,
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&error);
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if (error != RESAMPLER_ERR_SUCCESS) {
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return NS_ERROR_FAILURE;
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}
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}
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mSamplingRate = aSamplingRate;
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NS_ENSURE_TRUE(mSamplingRate > 0, NS_ERROR_FAILURE);
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int error = 0;
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mEncoder = opus_encoder_create(GetOutputSampleRate(), mChannels,
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OPUS_APPLICATION_AUDIO, &error);
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mInitialized = (error == OPUS_OK);
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if (mAudioBitrate) {
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opus_encoder_ctl(mEncoder, OPUS_SET_BITRATE(static_cast<int>(mAudioBitrate)));
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}
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mReentrantMonitor.NotifyAll();
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return error == OPUS_OK ? NS_OK : NS_ERROR_FAILURE;
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}
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int
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OpusTrackEncoder::GetOutputSampleRate()
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{
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return mResampler ? kOpusSamplingRate : mSamplingRate;
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}
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int
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OpusTrackEncoder::GetPacketDuration()
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{
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return GetOutputSampleRate() * kFrameDurationMs / 1000;
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}
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already_AddRefed<TrackMetadataBase>
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OpusTrackEncoder::GetMetadata()
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{
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PROFILER_LABEL("OpusTrackEncoder", "GetMetadata",
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js::ProfileEntry::Category::OTHER);
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{
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// Wait if mEncoder is not initialized.
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ReentrantMonitorAutoEnter mon(mReentrantMonitor);
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while (!mCanceled && !mInitialized) {
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mReentrantMonitor.Wait();
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}
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}
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if (mCanceled || mEncodingComplete) {
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return nullptr;
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}
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RefPtr<OpusMetadata> meta = new OpusMetadata();
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meta->mChannels = mChannels;
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meta->mSamplingFrequency = mSamplingRate;
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mLookahead = 0;
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int error = opus_encoder_ctl(mEncoder, OPUS_GET_LOOKAHEAD(&mLookahead));
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if (error != OPUS_OK) {
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mLookahead = 0;
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}
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// The ogg time stamping and pre-skip is always timed at 48000.
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SerializeOpusIdHeader(mChannels, mLookahead * (kOpusSamplingRate /
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GetOutputSampleRate()), mSamplingRate,
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&meta->mIdHeader);
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nsCString vendor;
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vendor.AppendASCII(opus_get_version_string());
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nsTArray<nsCString> comments;
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comments.AppendElement(NS_LITERAL_CSTRING("ENCODER=Mozilla" MOZ_APP_UA_VERSION));
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SerializeOpusCommentHeader(vendor, comments,
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&meta->mCommentHeader);
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return meta.forget();
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}
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nsresult
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OpusTrackEncoder::GetEncodedTrack(EncodedFrameContainer& aData)
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{
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PROFILER_LABEL("OpusTrackEncoder", "GetEncodedTrack",
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js::ProfileEntry::Category::OTHER);
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{
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ReentrantMonitorAutoEnter mon(mReentrantMonitor);
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// Wait until initialized or cancelled.
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while (!mCanceled && !mInitialized) {
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mReentrantMonitor.Wait();
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}
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if (mCanceled || mEncodingComplete) {
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return NS_ERROR_FAILURE;
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}
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}
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// calculation below depends on the truth that mInitialized is true.
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MOZ_ASSERT(mInitialized);
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bool wait = true;
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int result = 0;
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// Only wait once, then loop until we run out of packets of input data
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while (result >= 0 && !mEncodingComplete) {
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// re-sampled frames left last time which didn't fit into an Opus packet duration.
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const int framesLeft = mResampledLeftover.Length() / mChannels;
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// When framesLeft is 0, (GetPacketDuration() - framesLeft) is a multiple
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// of kOpusSamplingRate. There is not precision loss in the integer division
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// in computing framesToFetch. If frameLeft > 0, we need to add 1 to
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// framesToFetch to ensure there will be at least n frames after re-sampling.
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const int frameRoundUp = framesLeft ? 1 : 0;
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MOZ_ASSERT(GetPacketDuration() >= framesLeft);
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// Try to fetch m frames such that there will be n frames
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// where (n + frameLeft) >= GetPacketDuration() after re-sampling.
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const int framesToFetch = !mResampler ? GetPacketDuration()
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: (GetPacketDuration() - framesLeft) * mSamplingRate / kOpusSamplingRate
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+ frameRoundUp;
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{
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// Move all the samples from mRawSegment to mSourceSegment. We only hold
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// the monitor in this block.
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ReentrantMonitorAutoEnter mon(mReentrantMonitor);
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// Wait until enough raw data, end of stream or cancelled.
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while (!mCanceled && mRawSegment.GetDuration() +
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mSourceSegment.GetDuration() < framesToFetch &&
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!mEndOfStream) {
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if (wait) {
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mReentrantMonitor.Wait();
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wait = false;
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} else {
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goto done; // nested while's...
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}
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}
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if (mCanceled) {
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return NS_ERROR_FAILURE;
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}
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mSourceSegment.AppendFrom(&mRawSegment);
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// Pad |mLookahead| samples to the end of source stream to prevent lost of
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// original data, the pcm duration will be calculated at rate 48K later.
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if (mEndOfStream && !mEosSetInEncoder) {
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mEosSetInEncoder = true;
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mSourceSegment.AppendNullData(mLookahead);
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}
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}
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// Start encoding data.
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AutoTArray<AudioDataValue, 9600> pcm;
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pcm.SetLength(GetPacketDuration() * mChannels);
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AudioSegment::ChunkIterator iter(mSourceSegment);
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int frameCopied = 0;
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while (!iter.IsEnded() && frameCopied < framesToFetch) {
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AudioChunk chunk = *iter;
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// Chunk to the required frame size.
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int frameToCopy = chunk.GetDuration();
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if (frameCopied + frameToCopy > framesToFetch) {
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frameToCopy = framesToFetch - frameCopied;
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}
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if (!chunk.IsNull()) {
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// Append the interleaved data to the end of pcm buffer.
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AudioTrackEncoder::InterleaveTrackData(chunk, frameToCopy, mChannels,
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pcm.Elements() + frameCopied * mChannels);
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} else {
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memset(pcm.Elements() + frameCopied * mChannels, 0,
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frameToCopy * mChannels * sizeof(AudioDataValue));
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}
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frameCopied += frameToCopy;
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iter.Next();
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}
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RefPtr<EncodedFrame> audiodata = new EncodedFrame();
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audiodata->SetFrameType(EncodedFrame::OPUS_AUDIO_FRAME);
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int framesInPCM = frameCopied;
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if (mResampler) {
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AutoTArray<AudioDataValue, 9600> resamplingDest;
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// We want to consume all the input data, so we slightly oversize the
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// resampled data buffer so we can fit the output data in. We cannot really
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// predict the output frame count at each call.
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uint32_t outframes = frameCopied * kOpusSamplingRate / mSamplingRate + 1;
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uint32_t inframes = frameCopied;
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resamplingDest.SetLength(outframes * mChannels);
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#if MOZ_SAMPLE_TYPE_S16
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short* in = reinterpret_cast<short*>(pcm.Elements());
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short* out = reinterpret_cast<short*>(resamplingDest.Elements());
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speex_resampler_process_interleaved_int(mResampler, in, &inframes,
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out, &outframes);
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#else
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float* in = reinterpret_cast<float*>(pcm.Elements());
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float* out = reinterpret_cast<float*>(resamplingDest.Elements());
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speex_resampler_process_interleaved_float(mResampler, in, &inframes,
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out, &outframes);
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#endif
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MOZ_ASSERT(pcm.Length() >= mResampledLeftover.Length());
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PodCopy(pcm.Elements(), mResampledLeftover.Elements(),
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mResampledLeftover.Length());
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uint32_t outframesToCopy = std::min(outframes,
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static_cast<uint32_t>(GetPacketDuration() - framesLeft));
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MOZ_ASSERT(pcm.Length() - mResampledLeftover.Length() >=
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outframesToCopy * mChannels);
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PodCopy(pcm.Elements() + mResampledLeftover.Length(),
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resamplingDest.Elements(), outframesToCopy * mChannels);
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int frameLeftover = outframes - outframesToCopy;
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mResampledLeftover.SetLength(frameLeftover * mChannels);
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PodCopy(mResampledLeftover.Elements(),
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resamplingDest.Elements() + outframesToCopy * mChannels,
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mResampledLeftover.Length());
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// This is always at 48000Hz.
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framesInPCM = framesLeft + outframesToCopy;
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audiodata->SetDuration(framesInPCM);
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} else {
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// The ogg time stamping and pre-skip is always timed at 48000.
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audiodata->SetDuration(frameCopied * (kOpusSamplingRate / mSamplingRate));
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}
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// Remove the raw data which has been pulled to pcm buffer.
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// The value of frameCopied should equal to (or smaller than, if eos)
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// GetPacketDuration().
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mSourceSegment.RemoveLeading(frameCopied);
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// Has reached the end of input stream and all queued data has pulled for
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// encoding.
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if (mSourceSegment.GetDuration() == 0 && mEosSetInEncoder) {
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mEncodingComplete = true;
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LOG("[Opus] Done encoding.");
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}
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MOZ_ASSERT(mEosSetInEncoder || framesInPCM == GetPacketDuration());
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// Append null data to pcm buffer if the leftover data is not enough for
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// opus encoder.
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if (framesInPCM < GetPacketDuration() && mEosSetInEncoder) {
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PodZero(pcm.Elements() + framesInPCM * mChannels,
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(GetPacketDuration() - framesInPCM) * mChannels);
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}
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nsTArray<uint8_t> frameData;
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// Encode the data with Opus Encoder.
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frameData.SetLength(MAX_DATA_BYTES);
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// result is returned as opus error code if it is negative.
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result = 0;
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#ifdef MOZ_SAMPLE_TYPE_S16
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const opus_int16* pcmBuf = static_cast<opus_int16*>(pcm.Elements());
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result = opus_encode(mEncoder, pcmBuf, GetPacketDuration(),
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frameData.Elements(), MAX_DATA_BYTES);
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#else
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const float* pcmBuf = static_cast<float*>(pcm.Elements());
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result = opus_encode_float(mEncoder, pcmBuf, GetPacketDuration(),
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frameData.Elements(), MAX_DATA_BYTES);
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#endif
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frameData.SetLength(result >= 0 ? result : 0);
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if (result < 0) {
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LOG("[Opus] Fail to encode data! Result: %s.", opus_strerror(result));
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}
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if (mEncodingComplete) {
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if (mResampler) {
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speex_resampler_destroy(mResampler);
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mResampler = nullptr;
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}
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mResampledLeftover.SetLength(0);
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}
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audiodata->SwapInFrameData(frameData);
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// timestamp should be the time of the first sample
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audiodata->SetTimeStamp(mOutputTimeStamp);
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mOutputTimeStamp += FramesToUsecs(GetPacketDuration(), kOpusSamplingRate).value();
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LOG("[Opus] mOutputTimeStamp %lld.",mOutputTimeStamp);
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aData.AppendEncodedFrame(audiodata);
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}
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done:
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return result >= 0 ? NS_OK : NS_ERROR_FAILURE;
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}
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} // namespace mozilla
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