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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
commit6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch)
treea68f146d7fa01f0134297619fbe7e33db084e0aa /dom/media/platforms/apple/AppleATDecoder.cpp
parentInitial commit. (diff)
downloadthunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz
thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'dom/media/platforms/apple/AppleATDecoder.cpp')
-rw-r--r--dom/media/platforms/apple/AppleATDecoder.cpp672
1 files changed, 672 insertions, 0 deletions
diff --git a/dom/media/platforms/apple/AppleATDecoder.cpp b/dom/media/platforms/apple/AppleATDecoder.cpp
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--- /dev/null
+++ b/dom/media/platforms/apple/AppleATDecoder.cpp
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+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* 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 "AppleATDecoder.h"
+#include "Adts.h"
+#include "AppleUtils.h"
+#include "MP4Decoder.h"
+#include "MediaInfo.h"
+#include "VideoUtils.h"
+#include "mozilla/Logging.h"
+#include "mozilla/SyncRunnable.h"
+#include "mozilla/UniquePtr.h"
+#include "nsTArray.h"
+
+#define LOG(...) DDMOZ_LOG(sPDMLog, mozilla::LogLevel::Debug, __VA_ARGS__)
+#define LOGEX(_this, ...) \
+ DDMOZ_LOGEX(_this, sPDMLog, mozilla::LogLevel::Debug, __VA_ARGS__)
+#define FourCC2Str(n) \
+ ((char[5]){(char)(n >> 24), (char)(n >> 16), (char)(n >> 8), (char)(n), 0})
+
+namespace mozilla {
+
+AppleATDecoder::AppleATDecoder(const AudioInfo& aConfig)
+ : mConfig(aConfig),
+ mFileStreamError(false),
+ mConverter(nullptr),
+ mOutputFormat(),
+ mStream(nullptr),
+ mParsedFramesForAACMagicCookie(0),
+ mErrored(false) {
+ MOZ_COUNT_CTOR(AppleATDecoder);
+ LOG("Creating Apple AudioToolbox decoder");
+ LOG("Audio Decoder configuration: %s %d Hz %d channels %d bits per channel",
+ mConfig.mMimeType.get(), mConfig.mRate, mConfig.mChannels,
+ mConfig.mBitDepth);
+
+ if (mConfig.mMimeType.EqualsLiteral("audio/mpeg")) {
+ mFormatID = kAudioFormatMPEGLayer3;
+ } else if (mConfig.mMimeType.EqualsLiteral("audio/mp4a-latm")) {
+ mFormatID = kAudioFormatMPEG4AAC;
+ if (aConfig.mCodecSpecificConfig.is<AacCodecSpecificData>()) {
+ const AacCodecSpecificData& aacCodecSpecificData =
+ aConfig.mCodecSpecificConfig.as<AacCodecSpecificData>();
+ mEncoderDelay = aacCodecSpecificData.mEncoderDelayFrames;
+ mTotalMediaFrames = aacCodecSpecificData.mMediaFrameCount;
+ LOG("AppleATDecoder (aac), found encoder delay (%" PRIu32
+ ") and total frame count (%" PRIu64 ") in codec-specific side data",
+ mEncoderDelay, mTotalMediaFrames);
+ }
+ } else {
+ mFormatID = 0;
+ }
+}
+
+AppleATDecoder::~AppleATDecoder() {
+ MOZ_COUNT_DTOR(AppleATDecoder);
+ MOZ_ASSERT(!mConverter);
+}
+
+RefPtr<MediaDataDecoder::InitPromise> AppleATDecoder::Init() {
+ if (!mFormatID) {
+ return InitPromise::CreateAndReject(
+ MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
+ RESULT_DETAIL("Non recognised format")),
+ __func__);
+ }
+ mThread = GetCurrentSerialEventTarget();
+
+ return InitPromise::CreateAndResolve(TrackType::kAudioTrack, __func__);
+}
+
+RefPtr<MediaDataDecoder::FlushPromise> AppleATDecoder::Flush() {
+ MOZ_ASSERT(mThread->IsOnCurrentThread());
+ LOG("Flushing AudioToolbox AAC decoder");
+ mQueuedSamples.Clear();
+ mDecodedSamples.Clear();
+
+ if (mConverter) {
+ OSStatus rv = AudioConverterReset(mConverter);
+ if (rv) {
+ LOG("Error %d resetting AudioConverter", static_cast<int>(rv));
+ }
+ }
+ if (mErrored) {
+ mParsedFramesForAACMagicCookie = 0;
+ mMagicCookie.Clear();
+ ProcessShutdown();
+ mErrored = false;
+ }
+ return FlushPromise::CreateAndResolve(true, __func__);
+}
+
+RefPtr<MediaDataDecoder::DecodePromise> AppleATDecoder::Drain() {
+ MOZ_ASSERT(mThread->IsOnCurrentThread());
+ LOG("Draining AudioToolbox AAC decoder");
+ return DecodePromise::CreateAndResolve(DecodedData(), __func__);
+}
+
+RefPtr<ShutdownPromise> AppleATDecoder::Shutdown() {
+ // mThread may not be set if Init hasn't been called first.
+ MOZ_ASSERT(!mThread || mThread->IsOnCurrentThread());
+ ProcessShutdown();
+ return ShutdownPromise::CreateAndResolve(true, __func__);
+}
+
+void AppleATDecoder::ProcessShutdown() {
+ // mThread may not be set if Init hasn't been called first.
+ MOZ_ASSERT(!mThread || mThread->IsOnCurrentThread());
+
+ if (mStream) {
+ OSStatus rv = AudioFileStreamClose(mStream);
+ if (rv) {
+ LOG("error %d disposing of AudioFileStream", static_cast<int>(rv));
+ return;
+ }
+ mStream = nullptr;
+ }
+
+ if (mConverter) {
+ LOG("Shutdown: Apple AudioToolbox AAC decoder");
+ OSStatus rv = AudioConverterDispose(mConverter);
+ if (rv) {
+ LOG("error %d disposing of AudioConverter", static_cast<int>(rv));
+ }
+ mConverter = nullptr;
+ }
+}
+
+nsCString AppleATDecoder::GetCodecName() const {
+ switch (mFormatID) {
+ case kAudioFormatMPEGLayer3:
+ return "mp3"_ns;
+ case kAudioFormatMPEG4AAC:
+ return "aac"_ns;
+ default:
+ return "unknown"_ns;
+ }
+}
+
+struct PassthroughUserData {
+ UInt32 mChannels;
+ UInt32 mDataSize;
+ const void* mData;
+ AudioStreamPacketDescription mPacket;
+};
+
+// Error value we pass through the decoder to signal that nothing
+// has gone wrong during decoding and we're done processing the packet.
+const uint32_t kNoMoreDataErr = 'MOAR';
+
+static OSStatus _PassthroughInputDataCallback(
+ AudioConverterRef aAudioConverter, UInt32* aNumDataPackets /* in/out */,
+ AudioBufferList* aData /* in/out */,
+ AudioStreamPacketDescription** aPacketDesc, void* aUserData) {
+ PassthroughUserData* userData = (PassthroughUserData*)aUserData;
+ if (!userData->mDataSize) {
+ *aNumDataPackets = 0;
+ return kNoMoreDataErr;
+ }
+
+ if (aPacketDesc) {
+ userData->mPacket.mStartOffset = 0;
+ userData->mPacket.mVariableFramesInPacket = 0;
+ userData->mPacket.mDataByteSize = userData->mDataSize;
+ *aPacketDesc = &userData->mPacket;
+ }
+
+ aData->mBuffers[0].mNumberChannels = userData->mChannels;
+ aData->mBuffers[0].mDataByteSize = userData->mDataSize;
+ aData->mBuffers[0].mData = const_cast<void*>(userData->mData);
+
+ // No more data to provide following this run.
+ userData->mDataSize = 0;
+
+ return noErr;
+}
+
+RefPtr<MediaDataDecoder::DecodePromise> AppleATDecoder::Decode(
+ MediaRawData* aSample) {
+ MOZ_ASSERT(mThread->IsOnCurrentThread());
+ LOG("mp4 input sample pts=%s duration=%s %s %llu bytes audio",
+ aSample->mTime.ToString().get(), aSample->GetEndTime().ToString().get(),
+ aSample->mKeyframe ? " keyframe" : "",
+ (unsigned long long)aSample->Size());
+
+ MediaResult rv = NS_OK;
+ if (!mConverter) {
+ rv = SetupDecoder(aSample);
+ if (rv != NS_OK && rv != NS_ERROR_NOT_INITIALIZED) {
+ return DecodePromise::CreateAndReject(rv, __func__);
+ }
+ }
+
+ mQueuedSamples.AppendElement(aSample);
+
+ if (rv == NS_OK) {
+ for (size_t i = 0; i < mQueuedSamples.Length(); i++) {
+ rv = DecodeSample(mQueuedSamples[i]);
+ if (NS_FAILED(rv)) {
+ mErrored = true;
+ return DecodePromise::CreateAndReject(rv, __func__);
+ }
+ }
+ mQueuedSamples.Clear();
+ }
+
+ DecodedData results = std::move(mDecodedSamples);
+ mDecodedSamples = DecodedData();
+ return DecodePromise::CreateAndResolve(std::move(results), __func__);
+}
+
+MediaResult AppleATDecoder::DecodeSample(MediaRawData* aSample) {
+ MOZ_ASSERT(mThread->IsOnCurrentThread());
+
+ // Array containing the queued decoded audio frames, about to be output.
+ nsTArray<AudioDataValue> outputData;
+ UInt32 channels = mOutputFormat.mChannelsPerFrame;
+ // Pick a multiple of the frame size close to a power of two
+ // for efficient allocation. We're mainly using this decoder to decode AAC,
+ // that has packets of 1024 audio frames.
+ const uint32_t MAX_AUDIO_FRAMES = 1024;
+ const uint32_t maxDecodedSamples = MAX_AUDIO_FRAMES * channels;
+
+ // Descriptions for _decompressed_ audio packets. ignored.
+ auto packets = MakeUnique<AudioStreamPacketDescription[]>(MAX_AUDIO_FRAMES);
+
+ // This API insists on having packets spoon-fed to it from a callback.
+ // This structure exists only to pass our state.
+ PassthroughUserData userData = {channels, (UInt32)aSample->Size(),
+ aSample->Data()};
+
+ // Decompressed audio buffer
+ AlignedAudioBuffer decoded(maxDecodedSamples);
+ if (!decoded) {
+ return NS_ERROR_OUT_OF_MEMORY;
+ }
+
+ do {
+ AudioBufferList decBuffer;
+ decBuffer.mNumberBuffers = 1;
+ decBuffer.mBuffers[0].mNumberChannels = channels;
+ decBuffer.mBuffers[0].mDataByteSize =
+ maxDecodedSamples * sizeof(AudioDataValue);
+ decBuffer.mBuffers[0].mData = decoded.get();
+
+ // in: the max number of packets we can handle from the decoder.
+ // out: the number of packets the decoder is actually returning.
+ UInt32 numFrames = MAX_AUDIO_FRAMES;
+
+ OSStatus rv = AudioConverterFillComplexBuffer(
+ mConverter, _PassthroughInputDataCallback, &userData,
+ &numFrames /* in/out */, &decBuffer, packets.get());
+
+ if (rv && rv != kNoMoreDataErr) {
+ LOG("Error decoding audio sample: %d\n", static_cast<int>(rv));
+ return MediaResult(
+ NS_ERROR_DOM_MEDIA_DECODE_ERR,
+ RESULT_DETAIL("Error decoding audio sample: %d @ %s",
+ static_cast<int>(rv), aSample->mTime.ToString().get()));
+ }
+
+ if (numFrames) {
+ AudioDataValue* outputFrames = decoded.get();
+ outputData.AppendElements(outputFrames, numFrames * channels);
+ }
+
+ if (rv == kNoMoreDataErr) {
+ break;
+ }
+ } while (true);
+
+ if (outputData.IsEmpty()) {
+ return NS_OK;
+ }
+
+ size_t numFrames = outputData.Length() / channels;
+ int rate = mOutputFormat.mSampleRate;
+ media::TimeUnit duration(numFrames, rate);
+ if (!duration.IsValid()) {
+ NS_WARNING("Invalid count of accumulated audio samples");
+ return MediaResult(
+ NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
+ RESULT_DETAIL(
+ "Invalid count of accumulated audio samples: num:%llu rate:%d",
+ uint64_t(numFrames), rate));
+ }
+
+ LOG("Decoded audio packet [%s, %s] (duration: %s)\n",
+ aSample->mTime.ToString().get(), aSample->GetEndTime().ToString().get(),
+ duration.ToString().get());
+
+ AudioSampleBuffer data(outputData.Elements(), outputData.Length());
+ if (!data.Data()) {
+ return NS_ERROR_OUT_OF_MEMORY;
+ }
+ if (mChannelLayout && !mAudioConverter) {
+ AudioConfig in(*mChannelLayout, channels, rate);
+ AudioConfig out(AudioConfig::ChannelLayout::SMPTEDefault(*mChannelLayout),
+ channels, rate);
+ mAudioConverter = MakeUnique<AudioConverter>(in, out);
+ }
+ if (mAudioConverter && mChannelLayout && mChannelLayout->IsValid()) {
+ MOZ_ASSERT(mAudioConverter->CanWorkInPlace());
+ data = mAudioConverter->Process(std::move(data));
+ }
+
+ RefPtr<AudioData> audio = new AudioData(
+ aSample->mOffset, aSample->mTime, data.Forget(), channels, rate,
+ mChannelLayout && mChannelLayout->IsValid()
+ ? mChannelLayout->Map()
+ : AudioConfig::ChannelLayout::UNKNOWN_MAP);
+ MOZ_DIAGNOSTIC_ASSERT(duration == audio->mDuration, "must be equal");
+ mDecodedSamples.AppendElement(std::move(audio));
+ return NS_OK;
+}
+
+MediaResult AppleATDecoder::GetInputAudioDescription(
+ AudioStreamBasicDescription& aDesc, const nsTArray<uint8_t>& aExtraData) {
+ MOZ_ASSERT(mThread->IsOnCurrentThread());
+
+ // Request the properties from CoreAudio using the codec magic cookie
+ AudioFormatInfo formatInfo;
+ PodZero(&formatInfo.mASBD);
+ formatInfo.mASBD.mFormatID = mFormatID;
+ if (mFormatID == kAudioFormatMPEG4AAC) {
+ formatInfo.mASBD.mFormatFlags = mConfig.mExtendedProfile;
+ }
+ formatInfo.mMagicCookieSize = aExtraData.Length();
+ formatInfo.mMagicCookie = aExtraData.Elements();
+
+ UInt32 formatListSize;
+ // Attempt to retrieve the default format using
+ // kAudioFormatProperty_FormatInfo method.
+ // This method only retrieves the FramesPerPacket information required
+ // by the decoder, which depends on the codec type and profile.
+ aDesc.mFormatID = mFormatID;
+ aDesc.mChannelsPerFrame = mConfig.mChannels;
+ aDesc.mSampleRate = mConfig.mRate;
+ UInt32 inputFormatSize = sizeof(aDesc);
+ OSStatus rv = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL,
+ &inputFormatSize, &aDesc);
+ if (NS_WARN_IF(rv)) {
+ return MediaResult(
+ NS_ERROR_FAILURE,
+ RESULT_DETAIL("Unable to get format info:%d", int32_t(rv)));
+ }
+
+ // If any of the methods below fail, we will return the default format as
+ // created using kAudioFormatProperty_FormatInfo above.
+ rv = AudioFormatGetPropertyInfo(kAudioFormatProperty_FormatList,
+ sizeof(formatInfo), &formatInfo,
+ &formatListSize);
+ if (rv || (formatListSize % sizeof(AudioFormatListItem))) {
+ return NS_OK;
+ }
+ size_t listCount = formatListSize / sizeof(AudioFormatListItem);
+ auto formatList = MakeUnique<AudioFormatListItem[]>(listCount);
+
+ rv = AudioFormatGetProperty(kAudioFormatProperty_FormatList,
+ sizeof(formatInfo), &formatInfo, &formatListSize,
+ formatList.get());
+ if (rv) {
+ return NS_OK;
+ }
+ LOG("found %zu available audio stream(s)",
+ formatListSize / sizeof(AudioFormatListItem));
+ // Get the index number of the first playable format.
+ // This index number will be for the highest quality layer the platform
+ // is capable of playing.
+ UInt32 itemIndex;
+ UInt32 indexSize = sizeof(itemIndex);
+ rv = AudioFormatGetProperty(kAudioFormatProperty_FirstPlayableFormatFromList,
+ formatListSize, formatList.get(), &indexSize,
+ &itemIndex);
+ if (rv) {
+ return NS_OK;
+ }
+
+ aDesc = formatList[itemIndex].mASBD;
+
+ return NS_OK;
+}
+
+AudioConfig::Channel ConvertChannelLabel(AudioChannelLabel id) {
+ switch (id) {
+ case kAudioChannelLabel_Left:
+ return AudioConfig::CHANNEL_FRONT_LEFT;
+ case kAudioChannelLabel_Right:
+ return AudioConfig::CHANNEL_FRONT_RIGHT;
+ case kAudioChannelLabel_Mono:
+ case kAudioChannelLabel_Center:
+ return AudioConfig::CHANNEL_FRONT_CENTER;
+ case kAudioChannelLabel_LFEScreen:
+ return AudioConfig::CHANNEL_LFE;
+ case kAudioChannelLabel_LeftSurround:
+ return AudioConfig::CHANNEL_SIDE_LEFT;
+ case kAudioChannelLabel_RightSurround:
+ return AudioConfig::CHANNEL_SIDE_RIGHT;
+ case kAudioChannelLabel_CenterSurround:
+ return AudioConfig::CHANNEL_BACK_CENTER;
+ case kAudioChannelLabel_RearSurroundLeft:
+ return AudioConfig::CHANNEL_BACK_LEFT;
+ case kAudioChannelLabel_RearSurroundRight:
+ return AudioConfig::CHANNEL_BACK_RIGHT;
+ default:
+ return AudioConfig::CHANNEL_INVALID;
+ }
+}
+
+// Will set mChannelLayout if a channel layout could properly be identified
+// and is supported.
+nsresult AppleATDecoder::SetupChannelLayout() {
+ MOZ_ASSERT(mThread->IsOnCurrentThread());
+
+ // Determine the channel layout.
+ UInt32 propertySize;
+ UInt32 size;
+ OSStatus status = AudioConverterGetPropertyInfo(
+ mConverter, kAudioConverterOutputChannelLayout, &propertySize, NULL);
+ if (status || !propertySize) {
+ LOG("Couldn't get channel layout property (%s)", FourCC2Str(status));
+ return NS_ERROR_FAILURE;
+ }
+
+ auto data = MakeUnique<uint8_t[]>(propertySize);
+ size = propertySize;
+ status = AudioConverterGetProperty(
+ mConverter, kAudioConverterInputChannelLayout, &size, data.get());
+ if (status || size != propertySize) {
+ LOG("Couldn't get channel layout property (%s)", FourCC2Str(status));
+ return NS_ERROR_FAILURE;
+ }
+
+ AudioChannelLayout* layout =
+ reinterpret_cast<AudioChannelLayout*>(data.get());
+ AudioChannelLayoutTag tag = layout->mChannelLayoutTag;
+
+ // if tag is kAudioChannelLayoutTag_UseChannelDescriptions then the structure
+ // directly contains the the channel layout mapping.
+ // If tag is kAudioChannelLayoutTag_UseChannelBitmap then the layout will
+ // be defined via the bitmap and can be retrieved using
+ // kAudioFormatProperty_ChannelLayoutForBitmap property.
+ // Otherwise the tag itself describes the layout.
+ if (tag != kAudioChannelLayoutTag_UseChannelDescriptions) {
+ AudioFormatPropertyID property =
+ tag == kAudioChannelLayoutTag_UseChannelBitmap
+ ? kAudioFormatProperty_ChannelLayoutForBitmap
+ : kAudioFormatProperty_ChannelLayoutForTag;
+
+ if (property == kAudioFormatProperty_ChannelLayoutForBitmap) {
+ status = AudioFormatGetPropertyInfo(
+ property, sizeof(UInt32), &layout->mChannelBitmap, &propertySize);
+ } else {
+ status = AudioFormatGetPropertyInfo(
+ property, sizeof(AudioChannelLayoutTag), &tag, &propertySize);
+ }
+ if (status || !propertySize) {
+ LOG("Couldn't get channel layout property info (%s:%s)",
+ FourCC2Str(property), FourCC2Str(status));
+ return NS_ERROR_FAILURE;
+ }
+ data = MakeUnique<uint8_t[]>(propertySize);
+ layout = reinterpret_cast<AudioChannelLayout*>(data.get());
+ size = propertySize;
+
+ if (property == kAudioFormatProperty_ChannelLayoutForBitmap) {
+ status = AudioFormatGetProperty(property, sizeof(UInt32),
+ &layout->mChannelBitmap, &size, layout);
+ } else {
+ status = AudioFormatGetProperty(property, sizeof(AudioChannelLayoutTag),
+ &tag, &size, layout);
+ }
+ if (status || size != propertySize) {
+ LOG("Couldn't get channel layout property (%s:%s)", FourCC2Str(property),
+ FourCC2Str(status));
+ return NS_ERROR_FAILURE;
+ }
+ // We have retrieved the channel layout from the tag or bitmap.
+ // We can now directly use the channel descriptions.
+ layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
+ }
+
+ if (layout->mNumberChannelDescriptions != mOutputFormat.mChannelsPerFrame) {
+ LOG("Not matching the original channel number");
+ return NS_ERROR_FAILURE;
+ }
+
+ AutoTArray<AudioConfig::Channel, 8> channels;
+ channels.SetLength(layout->mNumberChannelDescriptions);
+ for (uint32_t i = 0; i < layout->mNumberChannelDescriptions; i++) {
+ AudioChannelLabel id = layout->mChannelDescriptions[i].mChannelLabel;
+ AudioConfig::Channel channel = ConvertChannelLabel(id);
+ channels[i] = channel;
+ }
+ mChannelLayout = MakeUnique<AudioConfig::ChannelLayout>(
+ mOutputFormat.mChannelsPerFrame, channels.Elements());
+ return NS_OK;
+}
+
+MediaResult AppleATDecoder::SetupDecoder(MediaRawData* aSample) {
+ MOZ_ASSERT(mThread->IsOnCurrentThread());
+ static const uint32_t MAX_FRAMES = 2;
+
+ if (mFormatID == kAudioFormatMPEG4AAC && mConfig.mExtendedProfile == 2 &&
+ mParsedFramesForAACMagicCookie < MAX_FRAMES) {
+ // Check for implicit SBR signalling if stream is AAC-LC
+ // This will provide us with an updated magic cookie for use with
+ // GetInputAudioDescription.
+ if (NS_SUCCEEDED(GetImplicitAACMagicCookie(aSample)) &&
+ !mMagicCookie.Length()) {
+ // nothing found yet, will try again later
+ mParsedFramesForAACMagicCookie++;
+ return NS_ERROR_NOT_INITIALIZED;
+ }
+ // An error occurred, fallback to using default stream description
+ }
+
+ LOG("Initializing Apple AudioToolbox decoder");
+
+ // Should we try and use magic cookie data from the AAC data? We do this if
+ // - We have an AAC config &
+ // - We do not aleady have magic cookie data.
+ // Otherwise we just use the existing cookie (which may be empty).
+ bool shouldUseAacMagicCookie =
+ mConfig.mCodecSpecificConfig.is<AacCodecSpecificData>() &&
+ mMagicCookie.IsEmpty();
+
+ nsTArray<uint8_t>& magicCookie =
+ shouldUseAacMagicCookie
+ ? *mConfig.mCodecSpecificConfig.as<AacCodecSpecificData>()
+ .mEsDescriptorBinaryBlob
+ : mMagicCookie;
+ AudioStreamBasicDescription inputFormat;
+ PodZero(&inputFormat);
+
+ MediaResult rv = GetInputAudioDescription(inputFormat, magicCookie);
+ if (NS_FAILED(rv)) {
+ return rv;
+ }
+ // Fill in the output format manually.
+ PodZero(&mOutputFormat);
+ mOutputFormat.mFormatID = kAudioFormatLinearPCM;
+ mOutputFormat.mSampleRate = inputFormat.mSampleRate;
+ mOutputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
+#if defined(MOZ_SAMPLE_TYPE_FLOAT32)
+ mOutputFormat.mBitsPerChannel = 32;
+ mOutputFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat | 0;
+#elif defined(MOZ_SAMPLE_TYPE_S16)
+ mOutputFormat.mBitsPerChannel = 16;
+ mOutputFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | 0;
+#else
+# error Unknown audio sample type
+#endif
+ // Set up the decoder so it gives us one sample per frame
+ mOutputFormat.mFramesPerPacket = 1;
+ mOutputFormat.mBytesPerPacket = mOutputFormat.mBytesPerFrame =
+ mOutputFormat.mChannelsPerFrame * mOutputFormat.mBitsPerChannel / 8;
+
+ OSStatus status =
+ AudioConverterNew(&inputFormat, &mOutputFormat, &mConverter);
+ if (status) {
+ LOG("Error %d constructing AudioConverter", int(status));
+ mConverter = nullptr;
+ return MediaResult(
+ NS_ERROR_FAILURE,
+ RESULT_DETAIL("Error constructing AudioConverter:%d", int32_t(status)));
+ }
+
+ if (magicCookie.Length() && mFormatID == kAudioFormatMPEG4AAC) {
+ status = AudioConverterSetProperty(
+ mConverter, kAudioConverterDecompressionMagicCookie,
+ magicCookie.Length(), magicCookie.Elements());
+ if (status) {
+ LOG("Error setting AudioConverter AAC cookie:%d", int32_t(status));
+ ProcessShutdown();
+ return MediaResult(
+ NS_ERROR_FAILURE,
+ RESULT_DETAIL("Error setting AudioConverter AAC cookie:%d",
+ int32_t(status)));
+ }
+ }
+
+ if (NS_FAILED(SetupChannelLayout())) {
+ NS_WARNING("Couldn't retrieve channel layout, will use default layout");
+ }
+
+ return NS_OK;
+}
+
+static void _MetadataCallback(void* aAppleATDecoder, AudioFileStreamID aStream,
+ AudioFileStreamPropertyID aProperty,
+ UInt32* aFlags) {
+ AppleATDecoder* decoder = static_cast<AppleATDecoder*>(aAppleATDecoder);
+ MOZ_RELEASE_ASSERT(decoder->mThread->IsOnCurrentThread());
+
+ LOGEX(decoder, "MetadataCallback receiving: '%s'", FourCC2Str(aProperty));
+ if (aProperty == kAudioFileStreamProperty_MagicCookieData) {
+ UInt32 size;
+ Boolean writeable;
+ OSStatus rv =
+ AudioFileStreamGetPropertyInfo(aStream, aProperty, &size, &writeable);
+ if (rv) {
+ LOGEX(decoder, "Couldn't get property info for '%s' (%s)",
+ FourCC2Str(aProperty), FourCC2Str(rv));
+ decoder->mFileStreamError = true;
+ return;
+ }
+ auto data = MakeUnique<uint8_t[]>(size);
+ rv = AudioFileStreamGetProperty(aStream, aProperty, &size, data.get());
+ if (rv) {
+ LOGEX(decoder, "Couldn't get property '%s' (%s)", FourCC2Str(aProperty),
+ FourCC2Str(rv));
+ decoder->mFileStreamError = true;
+ return;
+ }
+ decoder->mMagicCookie.AppendElements(data.get(), size);
+ }
+}
+
+static void _SampleCallback(void* aSBR, UInt32 aNumBytes, UInt32 aNumPackets,
+ const void* aData,
+ AudioStreamPacketDescription* aPackets) {}
+
+nsresult AppleATDecoder::GetImplicitAACMagicCookie(
+ const MediaRawData* aSample) {
+ MOZ_ASSERT(mThread->IsOnCurrentThread());
+
+ // Prepend ADTS header to AAC audio.
+ RefPtr<MediaRawData> adtssample(aSample->Clone());
+ if (!adtssample) {
+ return NS_ERROR_OUT_OF_MEMORY;
+ }
+ int8_t frequency_index = Adts::GetFrequencyIndex(mConfig.mRate);
+
+ bool rv = Adts::ConvertSample(mConfig.mChannels, frequency_index,
+ mConfig.mProfile, adtssample);
+ if (!rv) {
+ NS_WARNING("Failed to apply ADTS header");
+ return NS_ERROR_FAILURE;
+ }
+ if (!mStream) {
+ OSStatus rv = AudioFileStreamOpen(this, _MetadataCallback, _SampleCallback,
+ kAudioFileAAC_ADTSType, &mStream);
+ if (rv) {
+ NS_WARNING("Couldn't open AudioFileStream");
+ return NS_ERROR_FAILURE;
+ }
+ }
+
+ OSStatus status = AudioFileStreamParseBytes(
+ mStream, adtssample->Size(), adtssample->Data(), 0 /* discontinuity */);
+ if (status) {
+ NS_WARNING("Couldn't parse sample");
+ }
+
+ if (status || mFileStreamError || mMagicCookie.Length()) {
+ // We have decoded a magic cookie or an error occurred as such
+ // we won't need the stream any longer.
+ AudioFileStreamClose(mStream);
+ mStream = nullptr;
+ }
+
+ return (mFileStreamError || status) ? NS_ERROR_FAILURE : NS_OK;
+}
+
+} // namespace mozilla
+
+#undef LOG
+#undef LOGEX