mirror of
https://gitlab.winehq.org/wine/wine-gecko.git
synced 2024-09-13 09:24:08 -07:00
508 lines
16 KiB
C++
508 lines
16 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim:set ts=2 sw=2 sts=2 et cindent: */
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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
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "AppleUtils.h"
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#include "MP4Reader.h"
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#include "MP4Decoder.h"
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#include "mp4_demuxer/Adts.h"
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#include "mp4_demuxer/DecoderData.h"
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#include "AppleATDecoder.h"
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#include "prlog.h"
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#ifdef PR_LOGGING
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PRLogModuleInfo* GetAppleMediaLog();
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#define LOG(...) PR_LOG(GetAppleMediaLog(), PR_LOG_DEBUG, (__VA_ARGS__))
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#else
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#define LOG(...)
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#endif
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#define FourCC2Str(n) ((char[5]){(char)(n >> 24), (char)(n >> 16), (char)(n >> 8), (char)(n), 0})
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namespace mozilla {
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AppleATDecoder::AppleATDecoder(const mp4_demuxer::AudioDecoderConfig& aConfig,
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MediaTaskQueue* aAudioTaskQueue,
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MediaDataDecoderCallback* aCallback)
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: mConfig(aConfig)
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, mFileStreamError(false)
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, mTaskQueue(aAudioTaskQueue)
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, mCallback(aCallback)
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, mConverter(nullptr)
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, mStream(nullptr)
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{
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MOZ_COUNT_CTOR(AppleATDecoder);
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LOG("Creating Apple AudioToolbox decoder");
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LOG("Audio Decoder configuration: %s %d Hz %d channels %d bits per channel",
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mConfig.mime_type,
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mConfig.samples_per_second,
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mConfig.channel_count,
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mConfig.bits_per_sample);
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if (!strcmp(mConfig.mime_type, "audio/mpeg")) {
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mFormatID = kAudioFormatMPEGLayer3;
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} else if (!strcmp(mConfig.mime_type, "audio/mp4a-latm")) {
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mFormatID = kAudioFormatMPEG4AAC;
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} else {
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mFormatID = 0;
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}
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}
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AppleATDecoder::~AppleATDecoder()
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{
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MOZ_COUNT_DTOR(AppleATDecoder);
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MOZ_ASSERT(!mConverter);
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}
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nsresult
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AppleATDecoder::Init()
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{
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if (!mFormatID) {
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NS_ERROR("Non recognised format");
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return NS_ERROR_FAILURE;
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}
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return NS_OK;
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}
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nsresult
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AppleATDecoder::Input(mp4_demuxer::MP4Sample* aSample)
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{
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LOG("mp4 input sample %p %lld us %lld pts%s %llu bytes audio",
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aSample,
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aSample->duration,
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aSample->composition_timestamp,
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aSample->is_sync_point ? " keyframe" : "",
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(unsigned long long)aSample->size);
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// Queue a task to perform the actual decoding on a separate thread.
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mTaskQueue->Dispatch(
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NS_NewRunnableMethodWithArg<nsAutoPtr<mp4_demuxer::MP4Sample>>(
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this,
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&AppleATDecoder::SubmitSample,
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nsAutoPtr<mp4_demuxer::MP4Sample>(aSample)));
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return NS_OK;
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}
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nsresult
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AppleATDecoder::Flush()
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{
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LOG("Flushing AudioToolbox AAC decoder");
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mTaskQueue->Flush();
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mQueuedSamples.Clear();
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OSStatus rv = AudioConverterReset(mConverter);
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if (rv) {
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LOG("Error %d resetting AudioConverter", rv);
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return NS_ERROR_FAILURE;
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}
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return NS_OK;
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}
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nsresult
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AppleATDecoder::Drain()
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{
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LOG("Draining AudioToolbox AAC decoder");
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mTaskQueue->AwaitIdle();
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mCallback->DrainComplete();
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return Flush();
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}
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nsresult
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AppleATDecoder::Shutdown()
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{
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LOG("Shutdown: Apple AudioToolbox AAC decoder");
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mQueuedSamples.Clear();
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OSStatus rv = AudioConverterDispose(mConverter);
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if (rv) {
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LOG("error %d disposing of AudioConverter", rv);
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return NS_ERROR_FAILURE;
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}
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mConverter = nullptr;
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if (mStream) {
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rv = AudioFileStreamClose(mStream);
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if (rv) {
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LOG("error %d disposing of AudioFileStream", rv);
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return NS_ERROR_FAILURE;
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}
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mStream = nullptr;
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}
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return NS_OK;
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}
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struct PassthroughUserData {
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UInt32 mChannels;
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UInt32 mDataSize;
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const void* mData;
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AudioStreamPacketDescription mPacket;
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};
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// Error value we pass through the decoder to signal that nothing
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// has gone wrong during decoding and we're done processing the packet.
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const uint32_t kNoMoreDataErr = 'MOAR';
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static OSStatus
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_PassthroughInputDataCallback(AudioConverterRef aAudioConverter,
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UInt32* aNumDataPackets /* in/out */,
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AudioBufferList* aData /* in/out */,
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AudioStreamPacketDescription** aPacketDesc,
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void* aUserData)
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{
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PassthroughUserData* userData = (PassthroughUserData*)aUserData;
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if (!userData->mDataSize) {
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*aNumDataPackets = 0;
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return kNoMoreDataErr;
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}
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LOG("AudioConverter wants %u packets of audio data\n", *aNumDataPackets);
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if (aPacketDesc) {
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userData->mPacket.mStartOffset = 0;
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userData->mPacket.mVariableFramesInPacket = 0;
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userData->mPacket.mDataByteSize = userData->mDataSize;
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*aPacketDesc = &userData->mPacket;
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}
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aData->mBuffers[0].mNumberChannels = userData->mChannels;
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aData->mBuffers[0].mDataByteSize = userData->mDataSize;
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aData->mBuffers[0].mData = const_cast<void*>(userData->mData);
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// No more data to provide following this run.
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userData->mDataSize = 0;
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return noErr;
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}
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void
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AppleATDecoder::SubmitSample(nsAutoPtr<mp4_demuxer::MP4Sample> aSample)
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{
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nsresult rv = NS_OK;
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if (!mConverter) {
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rv = SetupDecoder(aSample);
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if (rv != NS_OK && rv != NS_ERROR_NOT_INITIALIZED) {
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mCallback->Error();
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return;
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}
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}
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mQueuedSamples.AppendElement(aSample);
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if (rv == NS_OK) {
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for (size_t i = 0; i < mQueuedSamples.Length(); i++) {
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if (NS_FAILED(DecodeSample(mQueuedSamples[i]))) {
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mQueuedSamples.Clear();
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mCallback->Error();
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return;
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}
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}
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mQueuedSamples.Clear();
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}
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if (mTaskQueue->IsEmpty()) {
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mCallback->InputExhausted();
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}
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}
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nsresult
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AppleATDecoder::DecodeSample(mp4_demuxer::MP4Sample* aSample)
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{
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// Array containing the queued decoded audio frames, about to be output.
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nsTArray<AudioDataValue> outputData;
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UInt32 channels = mOutputFormat.mChannelsPerFrame;
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// Pick a multiple of the frame size close to a power of two
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// for efficient allocation.
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const uint32_t MAX_AUDIO_FRAMES = 128;
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const uint32_t maxDecodedSamples = MAX_AUDIO_FRAMES * channels;
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// Descriptions for _decompressed_ audio packets. ignored.
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nsAutoArrayPtr<AudioStreamPacketDescription>
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packets(new AudioStreamPacketDescription[MAX_AUDIO_FRAMES]);
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// This API insists on having packets spoon-fed to it from a callback.
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// This structure exists only to pass our state.
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PassthroughUserData userData =
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{ channels, (UInt32)aSample->size, aSample->data };
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// Decompressed audio buffer
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nsAutoArrayPtr<AudioDataValue> decoded(new AudioDataValue[maxDecodedSamples]);
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do {
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AudioBufferList decBuffer;
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decBuffer.mNumberBuffers = 1;
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decBuffer.mBuffers[0].mNumberChannels = channels;
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decBuffer.mBuffers[0].mDataByteSize =
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maxDecodedSamples * sizeof(AudioDataValue);
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decBuffer.mBuffers[0].mData = decoded.get();
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// in: the max number of packets we can handle from the decoder.
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// out: the number of packets the decoder is actually returning.
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UInt32 numFrames = MAX_AUDIO_FRAMES;
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OSStatus rv = AudioConverterFillComplexBuffer(mConverter,
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_PassthroughInputDataCallback,
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&userData,
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&numFrames /* in/out */,
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&decBuffer,
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packets.get());
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if (rv && rv != kNoMoreDataErr) {
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LOG("Error decoding audio stream: %d\n", rv);
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return NS_ERROR_FAILURE;
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}
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if (numFrames) {
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outputData.AppendElements(decoded.get(), numFrames * channels);
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LOG("%d frames decoded", numFrames);
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}
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if (rv == kNoMoreDataErr) {
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LOG("done processing compressed packet");
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break;
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}
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} while (true);
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if (outputData.IsEmpty()) {
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return NS_OK;
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}
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size_t numFrames = outputData.Length() / channels;
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int rate = mOutputFormat.mSampleRate;
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CheckedInt<Microseconds> duration = FramesToUsecs(numFrames, rate);
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if (!duration.isValid()) {
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NS_WARNING("Invalid count of accumulated audio samples");
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return NS_ERROR_FAILURE;
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}
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LOG("pushed audio at time %lfs; duration %lfs\n",
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(double)aSample->composition_timestamp / USECS_PER_S,
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(double)duration.value() / USECS_PER_S);
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nsAutoArrayPtr<AudioDataValue> data(new AudioDataValue[outputData.Length()]);
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PodCopy(data.get(), &outputData[0], outputData.Length());
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nsRefPtr<AudioData> audio = new AudioData(aSample->byte_offset,
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aSample->composition_timestamp,
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duration.value(),
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numFrames,
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data.forget(),
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channels,
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rate);
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mCallback->Output(audio);
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return NS_OK;
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}
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nsresult
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AppleATDecoder::GetInputAudioDescription(AudioStreamBasicDescription& aDesc,
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const mozilla::Vector<uint8_t>& aExtraData)
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{
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// Request the properties from CoreAudio using the codec magic cookie
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AudioFormatInfo formatInfo;
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PodZero(&formatInfo.mASBD);
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formatInfo.mASBD.mFormatID = mFormatID;
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if (mFormatID == kAudioFormatMPEG4AAC) {
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formatInfo.mASBD.mFormatFlags = mConfig.extended_profile;
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}
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formatInfo.mMagicCookieSize = aExtraData.length();
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formatInfo.mMagicCookie = aExtraData.begin();
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UInt32 formatListSize;
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// Attempt to retrieve the default format using
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// kAudioFormatProperty_FormatInfo method.
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// This method only retrieves the FramesPerPacket information required
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// by the decoder, which depends on the codec type and profile.
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aDesc.mFormatID = mFormatID;
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aDesc.mChannelsPerFrame = mConfig.channel_count;
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aDesc.mSampleRate = mConfig.samples_per_second;
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UInt32 inputFormatSize = sizeof(aDesc);
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OSStatus rv = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
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0,
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NULL,
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&inputFormatSize,
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&aDesc);
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if (NS_WARN_IF(rv)) {
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return NS_ERROR_FAILURE;
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}
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// If any of the methods below fail, we will return the default format as
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// created using kAudioFormatProperty_FormatInfo above.
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rv = AudioFormatGetPropertyInfo(kAudioFormatProperty_FormatList,
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sizeof(formatInfo),
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&formatInfo,
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&formatListSize);
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if (rv || (formatListSize % sizeof(AudioFormatListItem))) {
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return NS_OK;
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}
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size_t listCount = formatListSize / sizeof(AudioFormatListItem);
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nsAutoArrayPtr<AudioFormatListItem> formatList(
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new AudioFormatListItem[listCount]);
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rv = AudioFormatGetProperty(kAudioFormatProperty_FormatList,
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sizeof(formatInfo),
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&formatInfo,
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&formatListSize,
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formatList);
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if (rv) {
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return NS_OK;
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}
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LOG("found %u available audio stream(s)",
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formatListSize / sizeof(AudioFormatListItem));
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// Get the index number of the first playable format.
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// This index number will be for the highest quality layer the platform
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// is capable of playing.
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UInt32 itemIndex;
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UInt32 indexSize = sizeof(itemIndex);
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rv = AudioFormatGetProperty(kAudioFormatProperty_FirstPlayableFormatFromList,
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formatListSize,
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formatList,
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&indexSize,
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&itemIndex);
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if (rv) {
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return NS_OK;
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}
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aDesc = formatList[itemIndex].mASBD;
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return NS_OK;
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}
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nsresult
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AppleATDecoder::SetupDecoder(mp4_demuxer::MP4Sample* aSample)
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{
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if (mFormatID == kAudioFormatMPEG4AAC &&
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mConfig.extended_profile == 2) {
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// Check for implicit SBR signalling if stream is AAC-LC
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// This will provide us with an updated magic cookie for use with
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// GetInputAudioDescription.
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if (NS_SUCCEEDED(GetImplicitAACMagicCookie(aSample)) &&
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!mMagicCookie.length()) {
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// nothing found yet, will try again later
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return NS_ERROR_NOT_INITIALIZED;
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}
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// An error occurred, fallback to using default stream description
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}
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LOG("Initializing Apple AudioToolbox decoder");
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AudioStreamBasicDescription inputFormat;
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PodZero(&inputFormat);
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nsresult rv =
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GetInputAudioDescription(inputFormat,
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mMagicCookie.length() ?
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mMagicCookie : mConfig.extra_data);
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if (NS_FAILED(rv)) {
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return rv;
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}
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// Fill in the output format manually.
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PodZero(&mOutputFormat);
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mOutputFormat.mFormatID = kAudioFormatLinearPCM;
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mOutputFormat.mSampleRate = inputFormat.mSampleRate;
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mOutputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
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#if defined(MOZ_SAMPLE_TYPE_FLOAT32)
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mOutputFormat.mBitsPerChannel = 32;
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mOutputFormat.mFormatFlags =
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kLinearPCMFormatFlagIsFloat |
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0;
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#else
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# error Unknown audio sample type
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#endif
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// Set up the decoder so it gives us one sample per frame
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mOutputFormat.mFramesPerPacket = 1;
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mOutputFormat.mBytesPerPacket = mOutputFormat.mBytesPerFrame
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= mOutputFormat.mChannelsPerFrame * mOutputFormat.mBitsPerChannel / 8;
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OSStatus status = AudioConverterNew(&inputFormat, &mOutputFormat, &mConverter);
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if (status) {
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LOG("Error %d constructing AudioConverter", status);
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mConverter = nullptr;
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return NS_ERROR_FAILURE;
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}
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return NS_OK;
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}
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static void
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_MetadataCallback(void* aAppleATDecoder,
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AudioFileStreamID aStream,
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AudioFileStreamPropertyID aProperty,
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UInt32* aFlags)
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{
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AppleATDecoder* decoder = static_cast<AppleATDecoder*>(aAppleATDecoder);
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LOG("MetadataCallback receiving: '%s'", FourCC2Str(aProperty));
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if (aProperty == kAudioFileStreamProperty_MagicCookieData) {
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UInt32 size;
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Boolean writeable;
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OSStatus rv = AudioFileStreamGetPropertyInfo(aStream,
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aProperty,
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&size,
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&writeable);
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if (rv) {
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LOG("Couldn't get property info for '%s' (%s)",
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FourCC2Str(aProperty), FourCC2Str(rv));
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decoder->mFileStreamError = true;
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return;
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}
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nsAutoArrayPtr<uint8_t> data(new uint8_t[size]);
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rv = AudioFileStreamGetProperty(aStream, aProperty,
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&size, data);
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if (rv) {
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LOG("Couldn't get property '%s' (%s)",
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FourCC2Str(aProperty), FourCC2Str(rv));
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decoder->mFileStreamError = true;
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return;
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}
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decoder->mMagicCookie.append(data.get(), size);
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}
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}
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static void
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_SampleCallback(void* aSBR,
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UInt32 aNumBytes,
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UInt32 aNumPackets,
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const void* aData,
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AudioStreamPacketDescription* aPackets)
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{
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}
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nsresult
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AppleATDecoder::GetImplicitAACMagicCookie(const mp4_demuxer::MP4Sample* aSample)
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{
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// Prepend ADTS header to AAC audio.
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mp4_demuxer::MP4Sample adtssample(*aSample);
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bool rv = mp4_demuxer::Adts::ConvertSample(mConfig.channel_count,
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mConfig.frequency_index,
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mConfig.aac_profile,
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&adtssample);
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if (!rv) {
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NS_WARNING("Failed to apply ADTS header");
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return NS_ERROR_FAILURE;
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}
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if (!mStream) {
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OSStatus rv = AudioFileStreamOpen(this,
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_MetadataCallback,
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_SampleCallback,
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kAudioFileAAC_ADTSType,
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&mStream);
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if (rv) {
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NS_WARNING("Couldn't open AudioFileStream");
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return NS_ERROR_FAILURE;
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}
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}
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OSStatus status = AudioFileStreamParseBytes(mStream,
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adtssample.size,
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adtssample.data,
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0 /* discontinuity */);
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if (status) {
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NS_WARNING("Couldn't parse sample");
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}
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if (status || mFileStreamError || mMagicCookie.length()) {
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// We have decoded a magic cookie or an error occurred as such
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// we won't need the stream any longer.
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AudioFileStreamClose(mStream);
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mStream = nullptr;
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}
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return (mFileStreamError || status) ? NS_ERROR_FAILURE : NS_OK;
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}
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} // namespace mozilla
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