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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/platforms/apple/AppleATDecoder.cpp
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'dom/media/platforms/apple/AppleATDecoder.cpp')
-rw-r--r--dom/media/platforms/apple/AppleATDecoder.cpp722
1 files changed, 722 insertions, 0 deletions
diff --git a/dom/media/platforms/apple/AppleATDecoder.cpp b/dom/media/platforms/apple/AppleATDecoder.cpp
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+++ 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 "AppleUtils.h"
+#include "MP4Decoder.h"
+#include "mp4_demuxer/Adts.h"
+#include "MediaInfo.h"
+#include "AppleATDecoder.h"
+#include "mozilla/Logging.h"
+#include "mozilla/SyncRunnable.h"
+#include "mozilla/UniquePtr.h"
+
+#define LOG(...) MOZ_LOG(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,
+ TaskQueue* aTaskQueue,
+ MediaDataDecoderCallback* aCallback)
+ : mConfig(aConfig)
+ , mFileStreamError(false)
+ , mTaskQueue(aTaskQueue)
+ , mCallback(aCallback)
+ , mConverter(nullptr)
+ , mStream(nullptr)
+ , mIsFlushing(false)
+ , 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;
+ } else {
+ mFormatID = 0;
+ }
+}
+
+AppleATDecoder::~AppleATDecoder()
+{
+ MOZ_COUNT_DTOR(AppleATDecoder);
+ MOZ_ASSERT(!mConverter);
+}
+
+RefPtr<MediaDataDecoder::InitPromise>
+AppleATDecoder::Init()
+{
+ if (!mFormatID) {
+ NS_ERROR("Non recognised format");
+ return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
+ }
+
+ return InitPromise::CreateAndResolve(TrackType::kAudioTrack, __func__);
+}
+
+void
+AppleATDecoder::Input(MediaRawData* aSample)
+{
+ MOZ_ASSERT(mCallback->OnReaderTaskQueue());
+ LOG("mp4 input sample %p %lld us %lld pts%s %llu bytes audio",
+ aSample,
+ aSample->mDuration,
+ aSample->mTime,
+ aSample->mKeyframe ? " keyframe" : "",
+ (unsigned long long)aSample->Size());
+
+ // Queue a task to perform the actual decoding on a separate thread.
+ nsCOMPtr<nsIRunnable> runnable =
+ NewRunnableMethod<RefPtr<MediaRawData>>(
+ this,
+ &AppleATDecoder::SubmitSample,
+ RefPtr<MediaRawData>(aSample));
+ mTaskQueue->Dispatch(runnable.forget());
+}
+
+void
+AppleATDecoder::ProcessFlush()
+{
+ MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
+ mQueuedSamples.Clear();
+ if (mConverter) {
+ OSStatus rv = AudioConverterReset(mConverter);
+ if (rv) {
+ LOG("Error %d resetting AudioConverter", rv);
+ }
+ }
+ if (mErrored) {
+ mParsedFramesForAACMagicCookie = 0;
+ mMagicCookie.Clear();
+ ProcessShutdown();
+ mErrored = false;
+ }
+}
+
+void
+AppleATDecoder::Flush()
+{
+ MOZ_ASSERT(mCallback->OnReaderTaskQueue());
+ LOG("Flushing AudioToolbox AAC decoder");
+ mIsFlushing = true;
+ nsCOMPtr<nsIRunnable> runnable =
+ NewRunnableMethod(this, &AppleATDecoder::ProcessFlush);
+ SyncRunnable::DispatchToThread(mTaskQueue, runnable);
+ mIsFlushing = false;
+}
+
+void
+AppleATDecoder::Drain()
+{
+ MOZ_ASSERT(mCallback->OnReaderTaskQueue());
+ LOG("Draining AudioToolbox AAC decoder");
+ mTaskQueue->AwaitIdle();
+ mCallback->DrainComplete();
+ Flush();
+}
+
+void
+AppleATDecoder::Shutdown()
+{
+ MOZ_ASSERT(mCallback->OnReaderTaskQueue());
+ nsCOMPtr<nsIRunnable> runnable =
+ NewRunnableMethod(this, &AppleATDecoder::ProcessShutdown);
+ SyncRunnable::DispatchToThread(mTaskQueue, runnable);
+}
+
+void
+AppleATDecoder::ProcessShutdown()
+{
+ MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
+
+ if (mStream) {
+ OSStatus rv = AudioFileStreamClose(mStream);
+ if (rv) {
+ LOG("error %d disposing of AudioFileStream", rv);
+ return;
+ }
+ mStream = nullptr;
+ }
+
+ if (mConverter) {
+ LOG("Shutdown: Apple AudioToolbox AAC decoder");
+ OSStatus rv = AudioConverterDispose(mConverter);
+ if (rv) {
+ LOG("error %d disposing of AudioConverter", rv);
+ }
+ mConverter = nullptr;
+ }
+}
+
+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;
+}
+
+void
+AppleATDecoder::SubmitSample(MediaRawData* aSample)
+{
+ MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
+
+ if (mIsFlushing) {
+ return;
+ }
+
+ MediaResult rv = NS_OK;
+ if (!mConverter) {
+ rv = SetupDecoder(aSample);
+ if (rv != NS_OK && rv != NS_ERROR_NOT_INITIALIZED) {
+ mCallback->Error(rv);
+ return;
+ }
+ }
+
+ 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;
+ mCallback->Error(rv);
+ return;
+ }
+ }
+ mQueuedSamples.Clear();
+ }
+ mCallback->InputExhausted();
+}
+
+MediaResult
+AppleATDecoder::DecodeSample(MediaRawData* aSample)
+{
+ MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
+
+ // 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.
+ const uint32_t MAX_AUDIO_FRAMES = 128;
+ 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", rv);
+ return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
+ RESULT_DETAIL("Error decoding audio sample: %d @ %lld",
+ rv, aSample->mTime));
+ }
+
+ if (numFrames) {
+ outputData.AppendElements(decoded.get(), 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 = FramesToTimeUnit(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));
+ }
+
+#ifdef LOG_SAMPLE_DECODE
+ LOG("pushed audio at time %lfs; duration %lfs\n",
+ (double)aSample->mTime / USECS_PER_S,
+ duration.ToSeconds());
+#endif
+
+ AudioSampleBuffer data(outputData.Elements(), outputData.Length());
+ if (!data.Data()) {
+ return NS_ERROR_OUT_OF_MEMORY;
+ }
+ if (mChannelLayout && !mAudioConverter) {
+ AudioConfig in(*mChannelLayout.get(), rate);
+ AudioConfig out(channels, rate);
+ if (!in.IsValid() || !out.IsValid()) {
+ return MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
+ RESULT_DETAIL("Invalid audio config"));
+ }
+ mAudioConverter = MakeUnique<AudioConverter>(in, out);
+ }
+ if (mAudioConverter) {
+ MOZ_ASSERT(mAudioConverter->CanWorkInPlace());
+ data = mAudioConverter->Process(Move(data));
+ }
+
+ RefPtr<AudioData> audio = new AudioData(aSample->mOffset,
+ aSample->mTime,
+ duration.ToMicroseconds(),
+ numFrames,
+ data.Forget(),
+ channels,
+ rate);
+ mCallback->Output(audio);
+ return NS_OK;
+}
+
+MediaResult
+AppleATDecoder::GetInputAudioDescription(AudioStreamBasicDescription& aDesc,
+ const nsTArray<uint8_t>& aExtraData)
+{
+ MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
+
+ // 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:%lld", int64_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 %u 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_Mono:
+ return AudioConfig::CHANNEL_MONO;
+ case kAudioChannelLabel_Left:
+ return AudioConfig::CHANNEL_LEFT;
+ case kAudioChannelLabel_Right:
+ return AudioConfig::CHANNEL_RIGHT;
+ case kAudioChannelLabel_Center:
+ return AudioConfig::CHANNEL_CENTER;
+ case kAudioChannelLabel_LFEScreen:
+ return AudioConfig::CHANNEL_LFE;
+ case kAudioChannelLabel_LeftSurround:
+ return AudioConfig::CHANNEL_LS;
+ case kAudioChannelLabel_RightSurround:
+ return AudioConfig::CHANNEL_RS;
+ case kAudioChannelLabel_CenterSurround:
+ return AudioConfig::CHANNEL_RCENTER;
+ case kAudioChannelLabel_RearSurroundLeft:
+ return AudioConfig::CHANNEL_RLS;
+ case kAudioChannelLabel_RearSurroundRight:
+ return AudioConfig::CHANNEL_RRS;
+ default:
+ return AudioConfig::CHANNEL_INVALID;
+ }
+}
+
+// Will set mChannelLayout if a channel layout could properly be identified
+// and is supported.
+nsresult
+AppleATDecoder::SetupChannelLayout()
+{
+ MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
+
+ // 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 > MAX_AUDIO_CHANNELS ||
+ layout->mNumberChannelDescriptions != mOutputFormat.mChannelsPerFrame) {
+ LOG("Nonsensical channel layout or not matching the original channel number");
+ return NS_ERROR_FAILURE;
+ }
+
+ AudioConfig::Channel channels[MAX_AUDIO_CHANNELS];
+ 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);
+ return NS_OK;
+}
+
+MediaResult
+AppleATDecoder::SetupDecoder(MediaRawData* aSample)
+{
+ MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
+ 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");
+
+ AudioStreamBasicDescription inputFormat;
+ PodZero(&inputFormat);
+ MediaResult rv =
+ GetInputAudioDescription(inputFormat,
+ mMagicCookie.Length() ?
+ mMagicCookie : *mConfig.mExtraData);
+ 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", status);
+ mConverter = nullptr;
+ return MediaResult(
+ NS_ERROR_FAILURE,
+ RESULT_DETAIL("Error constructing AudioConverter:%lld", int64_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);
+ LOG("MetadataCallback receiving: '%s'", FourCC2Str(aProperty));
+ if (aProperty == kAudioFileStreamProperty_MagicCookieData) {
+ UInt32 size;
+ Boolean writeable;
+ OSStatus rv = AudioFileStreamGetPropertyInfo(aStream,
+ aProperty,
+ &size,
+ &writeable);
+ if (rv) {
+ LOG("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) {
+ LOG("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(mTaskQueue->IsCurrentThreadIn());
+
+ // Prepend ADTS header to AAC audio.
+ RefPtr<MediaRawData> adtssample(aSample->Clone());
+ if (!adtssample) {
+ return NS_ERROR_OUT_OF_MEMORY;
+ }
+ int8_t frequency_index =
+ mp4_demuxer::Adts::GetFrequencyIndex(mConfig.mRate);
+
+ bool rv = mp4_demuxer::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