horok/firefox
1
0
Fork 0
Code Activity
firefox/dom/media/webcodecs/AudioData.cpp
Daniel Baumann 5e9a113729
Adding upstream version 140.0. 
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
2025-06-25 09:37:52 +02:00

727 lines
25 KiB
C++
Raw Permalink Blame History

/* -*- 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 "mozilla/Assertions.h"
#include "mozilla/CheckedInt.h"
#include "mozilla/Logging.h"
#include "mozilla/dom/AudioData.h"
#include "mozilla/dom/AudioDataBinding.h"
#include "mozilla/dom/BufferSourceBinding.h"
#include "mozilla/dom/Promise.h"
#include "mozilla/dom/StructuredCloneTags.h"
#include "nsStringFwd.h"
#include "nsFmtString.h"
#include <utility>
#include "AudioSampleFormat.h"
#include "WebCodecsUtils.h"
#include "js/StructuredClone.h"
#include "mozilla/Maybe.h"
#include "mozilla/Result.h"
extern mozilla::LazyLogModule gWebCodecsLog;
namespace mozilla::dom {
#define LOGD(fmt, ...) \
MOZ_LOG_FMT(gWebCodecsLog, LogLevel::Debug, fmt, ##__VA_ARGS__)
#define LOGE(fmt, ...) \
MOZ_LOG_FMT(gWebCodecsLog, LogLevel::Error, fmt, ##__VA_ARGS__)
[[nodiscard]] Result<Ok, nsCString> LogAndReturnErr(const char* aLiteral) {
MOZ_LOG(gWebCodecsLog, LogLevel::Debug, ("%s", aLiteral));
return Err(nsCString(aLiteral));
}
template <typename... Args>
[[nodiscard]] Result<Ok, nsCString> LogAndReturnErr(
fmt::format_string<Args...> aFmt, Args&&... aArgs) {
nsAutoCStringN<100> str;
str.AppendVfmt(aFmt, fmt::make_format_args(aArgs...));
MOZ_LOG(gWebCodecsLog, LogLevel::Debug, ("%s", str.get()));
return Err(str);
}
// Only needed for refcounted objects.
//
NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(AudioData)
NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioData)
tmp->CloseIfNeeded();
NS_IMPL_CYCLE_COLLECTION_UNLINK(mParent)
NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
NS_IMPL_CYCLE_COLLECTION_UNLINK_END
NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(AudioData)
NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mParent)
NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
NS_IMPL_CYCLE_COLLECTING_ADDREF(AudioData)
// AudioData should be released as soon as its refcount drops to zero,
// without waiting for async deletion by the cycle collector, since it may hold
// a large-size PCM buffer.
NS_IMPL_CYCLE_COLLECTING_RELEASE_WITH_LAST_RELEASE(AudioData, CloseIfNeeded())
NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(AudioData)
NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
NS_INTERFACE_MAP_ENTRY(nsISupports)
NS_INTERFACE_MAP_END
/*
* W3C Webcodecs AudioData implementation
*/
AudioData::AudioData(nsIGlobalObject* aParent,
const AudioDataSerializedData& aData)
: mParent(aParent),
mTimestamp(aData.mTimestamp),
mNumberOfChannels(aData.mNumberOfChannels),
mNumberOfFrames(aData.mNumberOfFrames),
mSampleRate(aData.mSampleRate),
mAudioSampleFormat(aData.mAudioSampleFormat),
// The resource is not copied, but referenced
mResource(aData.mResource) {
MOZ_ASSERT(mParent);
MOZ_ASSERT(mResource,
"Resource should always be present then receiving a transfer.");
}
AudioData::AudioData(const AudioData& aOther)
: mParent(aOther.mParent),
mTimestamp(aOther.mTimestamp),
mNumberOfChannels(aOther.mNumberOfChannels),
mNumberOfFrames(aOther.mNumberOfFrames),
mSampleRate(aOther.mSampleRate),
mAudioSampleFormat(aOther.mAudioSampleFormat),
// The resource is not copied, but referenced
mResource(aOther.mResource) {
MOZ_ASSERT(mParent);
}
Result<already_AddRefed<AudioDataResource>, nsresult>
AudioDataResource::Construct(const OwningAllowSharedBufferSource& aInit) {
FallibleTArray<uint8_t> copied;
uint8_t* rv = ProcessTypedArraysFixed(
aInit, [&](const Span<uint8_t>& aData) -> uint8_t* {
return copied.AppendElements(aData.Elements(), aData.Length(),
fallible);
});
if (!rv) {
LOGE("AudioDataResource::Ctor: OOM");
return Err(NS_ERROR_OUT_OF_MEMORY);
}
return MakeAndAddRef<AudioDataResource>(std::move(copied));
}
AudioData::AudioData(
nsIGlobalObject* aParent,
already_AddRefed<mozilla::dom::AudioDataResource> aResource,
const AudioDataInit& aInit)
: mParent(aParent),
mTimestamp(aInit.mTimestamp),
mNumberOfChannels(aInit.mNumberOfChannels),
mNumberOfFrames(aInit.mNumberOfFrames),
mSampleRate(aInit.mSampleRate),
mAudioSampleFormat(Some(aInit.mFormat)),
mResource(std::move(aResource)) {
MOZ_ASSERT(mParent);
}
AudioData::AudioData(
nsIGlobalObject* aParent,
already_AddRefed<mozilla::dom::AudioDataResource> aResource,
int64_t aTimestamp, uint32_t aNumberOfChannels, uint32_t aNumberOfFrames,
float aSampleRate, AudioSampleFormat aAudioSampleFormat)
: mParent(aParent),
mTimestamp(aTimestamp),
mNumberOfChannels(aNumberOfChannels),
mNumberOfFrames(aNumberOfFrames),
mSampleRate(aSampleRate),
mAudioSampleFormat(Some(aAudioSampleFormat)),
mResource(aResource) {
MOZ_ASSERT(mParent);
}
nsIGlobalObject* AudioData::GetParentObject() const {
AssertIsOnOwningThread();
return mParent.get();
}
JSObject* AudioData::WrapObject(JSContext* aCx,
JS::Handle<JSObject*> aGivenProto) {
AssertIsOnOwningThread();
return AudioData_Binding::Wrap(aCx, this, aGivenProto);
}
Result<Ok, nsCString> IsValidAudioDataInit(const AudioDataInit& aInit) {
// The sample rate is an uint32_t within Gecko
uint32_t integerSampleRate = SaturatingCast<uint32_t>(aInit.mSampleRate);
if (integerSampleRate == 0) {
return LogAndReturnErr("sampleRate must be positive");
}
if (aInit.mNumberOfFrames == 0) {
return LogAndReturnErr("mNumberOfFrames must be positive");
}
if (aInit.mNumberOfChannels == 0) {
return LogAndReturnErr("mNumberOfChannels must be positive");
}
CheckedInt<uint64_t> bytesNeeded = aInit.mNumberOfFrames;
bytesNeeded *= aInit.mNumberOfChannels;
bytesNeeded *= BytesPerSamples(aInit.mFormat);
if (!bytesNeeded.isValid()) {
return LogAndReturnErr(
FMT_STRING("Overflow when computing the number of bytes needed to hold "
"audio samples ({}*{}*{})"),
aInit.mNumberOfFrames, aInit.mNumberOfChannels,
BytesPerSamples(aInit.mFormat));
}
uint64_t arraySizeBytes = ProcessTypedArraysFixed(
aInit.mData, [&](const Span<uint8_t>& aData) -> uint64_t {
return aData.LengthBytes();
});
if (arraySizeBytes < bytesNeeded.value()) {
return LogAndReturnErr(
FMT_STRING("Array of size {} not big enough, should be at least {}"),
arraySizeBytes, bytesNeeded.value());
}
return Ok();
}
/* static */
already_AddRefed<AudioData> AudioData::Constructor(const GlobalObject& aGlobal,
const AudioDataInit& aInit,
ErrorResult& aRv) {
nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(aGlobal.GetAsSupports());
LOGD("[{}] AudioData(fmt: {}, rate: {}, ch: {}, ts: {})",
fmt::ptr(global.get()), GetEnumString(aInit.mFormat).get(),
aInit.mSampleRate, aInit.mNumberOfChannels, aInit.mTimestamp);
if (!global) {
LOGE("Global unavailable");
aRv.Throw(NS_ERROR_FAILURE);
return nullptr;
}
nsString errorMessage;
auto rv = IsValidAudioDataInit(aInit);
if (rv.isErr()) {
LOGD("AudioData::Constructor failure (IsValidAudioDataInit)");
aRv.ThrowTypeError(rv.inspectErr());
return nullptr;
}
auto resource = AudioDataResource::Construct(aInit.mData);
if (resource.isErr()) {
LOGD("AudioData::Constructor failure (OOM)");
aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
return nullptr;
}
return MakeAndAddRef<mozilla::dom::AudioData>(global, resource.unwrap(),
aInit);
}
// https://w3c.github.io/webcodecs/#dom-audiodata-format
Nullable<mozilla::dom::AudioSampleFormat> AudioData::GetFormat() const {
AssertIsOnOwningThread();
return MaybeToNullable(mAudioSampleFormat);
}
// https://w3c.github.io/webcodecs/#dom-audiodata-samplerate
float AudioData::SampleRate() const {
AssertIsOnOwningThread();
return mSampleRate;
}
// https://w3c.github.io/webcodecs/#dom-audiodata-numberofframes
uint32_t AudioData::NumberOfFrames() const {
AssertIsOnOwningThread();
return mNumberOfFrames;
}
// https://w3c.github.io/webcodecs/#dom-audiodata-numberofchannels
uint32_t AudioData::NumberOfChannels() const {
AssertIsOnOwningThread();
return mNumberOfChannels;
}
// https://w3c.github.io/webcodecs/#dom-audiodata-duration
uint64_t AudioData::Duration() const {
AssertIsOnOwningThread();
if (!mNumberOfFrames) {
return 0;
}
// The spec isn't clear in which direction to convert to integer.
// https://github.com/w3c/webcodecs/issues/726
return static_cast<uint64_t>(
static_cast<float>(USECS_PER_S * mNumberOfFrames) / mSampleRate);
}
// https://w3c.github.io/webcodecs/#dom-audiodata-timestamp
int64_t AudioData::Timestamp() const {
AssertIsOnOwningThread();
return mTimestamp;
}
struct CopyToSpec {
CopyToSpec(uint32_t aFrameCount, uint32_t aFrameOffset, uint32_t mPlaneIndex,
AudioSampleFormat aFormat)
: mFrameCount(aFrameCount),
mFrameOffset(aFrameOffset),
mPlaneIndex(mPlaneIndex),
mFormat(aFormat) {}
const uint32_t mFrameCount;
const uint32_t mFrameOffset;
const uint32_t mPlaneIndex;
const AudioSampleFormat mFormat;
};
bool IsInterleaved(const AudioSampleFormat& aFormat) {
switch (aFormat) {
case AudioSampleFormat::U8:
case AudioSampleFormat::S16:
case AudioSampleFormat::S32:
case AudioSampleFormat::F32:
return true;
case AudioSampleFormat::U8_planar:
case AudioSampleFormat::S16_planar:
case AudioSampleFormat::S32_planar:
case AudioSampleFormat::F32_planar:
return false;
};
MOZ_ASSERT_UNREACHABLE("Invalid enum value");
return false;
}
size_t AudioData::ComputeCopyElementCount(
const AudioDataCopyToOptions& aOptions, ErrorResult& aRv) {
// https://w3c.github.io/webcodecs/#compute-copy-element-count
// 1, 2
auto destFormat = mAudioSampleFormat;
if (aOptions.mFormat.WasPassed()) {
destFormat = OptionalToMaybe(aOptions.mFormat);
}
// 3, 4
MOZ_ASSERT(destFormat.isSome());
if (IsInterleaved(destFormat.value())) {
if (aOptions.mPlaneIndex > 0) {
auto msg = "Interleaved format, but plane index > 0"_ns;
LOGD("{}", msg.get());
aRv.ThrowRangeError(msg);
return 0;
}
} else {
if (aOptions.mPlaneIndex >= mNumberOfChannels) {
auto msg = nsFmtCString(FMT_STRING("Plane index {} greater or equal "
"than the number of channels {}"),
aOptions.mPlaneIndex, mNumberOfChannels);
LOGD("{}", msg.get());
aRv.ThrowRangeError(msg);
return 0;
}
}
// 5 -- conversion between all formats supported
// 6 -- all planes have the same number of frames, always
uint64_t frameCount = mNumberOfFrames;
// 7
if (aOptions.mFrameOffset >= frameCount) {
auto msg = nsFmtCString(
FMT_STRING("Frame offset of {} greater or equal than frame count {}"),
aOptions.mFrameOffset, frameCount);
LOGD("{}", msg.get());
aRv.ThrowRangeError(msg);
return 0;
}
// 8, 9
uint64_t copyFrameCount = frameCount - aOptions.mFrameOffset;
if (aOptions.mFrameCount.WasPassed()) {
if (aOptions.mFrameCount.Value() > copyFrameCount) {
auto msg = nsFmtCString(FMT_STRING("Passed copy frame count of {} "
"greater than available source frames "
"for copy of {}"),
aOptions.mFrameCount.Value(), copyFrameCount);
LOGD("{}", msg.get());
aRv.ThrowRangeError(msg);
return 0;
}
copyFrameCount = aOptions.mFrameCount.Value();
}
// 10, 11
uint64_t elementCount = copyFrameCount;
if (IsInterleaved(destFormat.value())) {
elementCount *= mNumberOfChannels;
}
return elementCount;
}
// https://w3c.github.io/webcodecs/#dom-audiodata-allocationsize
// This method returns an int, that can be zero in case of success or error.
// Caller should check aRv to determine success or error.
uint32_t AudioData::AllocationSize(const AudioDataCopyToOptions& aOptions,
ErrorResult& aRv) {
AssertIsOnOwningThread();
if (!mResource) {
auto msg = "allocationSize called on detached AudioData"_ns;
LOGD("{}", msg.get());
aRv.ThrowInvalidStateError(msg);
return 0;
}
size_t copyElementCount = ComputeCopyElementCount(aOptions, aRv);
if (aRv.Failed()) {
LOGD("AudioData::AllocationSize failure");
// ComputeCopyElementCount has set the exception type.
return 0;
}
Maybe<mozilla::dom::AudioSampleFormat> destFormat = mAudioSampleFormat;
if (aOptions.mFormat.WasPassed()) {
destFormat = OptionalToMaybe(aOptions.mFormat);
}
if (destFormat.isNothing()) {
auto msg = "AudioData has an unknown format"_ns;
LOGD("{}", msg.get());
// See https://github.com/w3c/webcodecs/issues/727 -- it isn't clear yet
// what to do here
aRv.ThrowRangeError(msg);
return 0;
}
CheckedInt<size_t> bytesPerSample = BytesPerSamples(destFormat.ref());
auto res = bytesPerSample * copyElementCount;
if (res.isValid()) {
return res.value();
}
aRv.ThrowRangeError("Allocation size too large");
return 0;
}
template <typename S, typename D>
void CopySamples(Span<S> aSource, Span<D> aDest, uint32_t aSourceChannelCount,
const AudioSampleFormat aSourceFormat,
const CopyToSpec& aCopyToSpec) {
if (IsInterleaved(aSourceFormat) && IsInterleaved(aCopyToSpec.mFormat)) {
MOZ_ASSERT(aCopyToSpec.mPlaneIndex == 0);
MOZ_ASSERT(aDest.Length() >= aCopyToSpec.mFrameCount);
MOZ_ASSERT(aSource.Length() - aCopyToSpec.mFrameOffset >=
aCopyToSpec.mFrameCount);
// This turns into a regular memcpy if the types are in fact equal
ConvertAudioSamples(aSource.data() + aCopyToSpec.mFrameOffset, aDest.data(),
aCopyToSpec.mFrameCount * aSourceChannelCount);
return;
}
if (IsInterleaved(aSourceFormat) && !IsInterleaved(aCopyToSpec.mFormat)) {
DebugOnly<size_t> sourceFrameCount = aSource.Length() / aSourceChannelCount;
MOZ_ASSERT(aDest.Length() >= aCopyToSpec.mFrameCount);
MOZ_ASSERT(aSource.Length() - aCopyToSpec.mFrameOffset >=
aCopyToSpec.mFrameCount);
// Interleaved to planar -- only copy samples of the correct channel to the
// destination
size_t readIndex = aCopyToSpec.mFrameOffset * aSourceChannelCount +
aCopyToSpec.mPlaneIndex;
for (size_t i = 0; i < aCopyToSpec.mFrameCount; i++) {
aDest[i] = ConvertAudioSample<D>(aSource[readIndex]);
readIndex += aSourceChannelCount;
}
return;
}
if (!IsInterleaved(aSourceFormat) && IsInterleaved(aCopyToSpec.mFormat)) {
// Planar to interleaved -- copy of all channels of the source into the
// destination buffer.
MOZ_ASSERT(aCopyToSpec.mPlaneIndex == 0);
MOZ_ASSERT(aDest.Length() >= aCopyToSpec.mFrameCount * aSourceChannelCount);
MOZ_ASSERT(aSource.Length() -
aCopyToSpec.mFrameOffset * aSourceChannelCount >=
aCopyToSpec.mFrameCount * aSourceChannelCount);
size_t writeIndex = 0;
// Scan the source linearly and put each sample at the right position in the
// destination interleaved buffer.
size_t readIndex = 0;
for (size_t channel = 0; channel < aSourceChannelCount; channel++) {
writeIndex = channel;
for (size_t i = 0; i < aCopyToSpec.mFrameCount; i++) {
aDest[writeIndex] = ConvertAudioSample<D>(aSource[readIndex]);
readIndex++;
writeIndex += aSourceChannelCount;
}
}
return;
}
if (!IsInterleaved(aSourceFormat) && !IsInterleaved(aCopyToSpec.mFormat)) {
// Planar to Planar / convert + copy from the right index in the source.
size_t framePerPlane = aSource.Length() / aSourceChannelCount;
size_t offset = aCopyToSpec.mPlaneIndex * framePerPlane;
MOZ_ASSERT(aDest.Length() >= aCopyToSpec.mFrameCount,
"Destination buffer too small");
MOZ_ASSERT(aSource.Length() >= offset + aCopyToSpec.mFrameCount,
"Source buffer too small");
for (uint32_t i = 0; i < aCopyToSpec.mFrameCount; i++) {
aDest[i] =
ConvertAudioSample<D>(aSource[offset + aCopyToSpec.mFrameOffset + i]);
}
}
}
nsCString AudioData::ToString() const {
if (!mResource) {
return nsCString("AudioData[detached]");
}
return nsFmtCString(FMT_STRING("AudioData[{} bytes {} {}Hz {} x {}ch]"),
mResource->Data().LengthBytes(),
GetEnumString(mAudioSampleFormat.value()).get(),
mSampleRate, mNumberOfFrames, mNumberOfChannels);
}
nsCString CopyToToString(size_t aDestBufSize,
const AudioDataCopyToOptions& aOptions) {
return nsFmtCString(
FMT_STRING(
"AudioDataCopyToOptions[data: {} bytes, {}, frame count: {}, frame "
"offset: {}, plane: {}]"),
aDestBufSize,
aOptions.mFormat.WasPassed()
? GetEnumString(aOptions.mFormat.Value()).get()
: "null",
aOptions.mFrameCount.WasPassed() ? aOptions.mFrameCount.Value() : 0,
aOptions.mFrameOffset, aOptions.mPlaneIndex);
}
using DataSpanType =
Variant<Span<uint8_t>, Span<int16_t>, Span<int32_t>, Span<float>>;
DataSpanType GetDataSpan(Span<uint8_t> aSpan, const AudioSampleFormat aFormat) {
const size_t Length = aSpan.Length() / BytesPerSamples(aFormat);
// TODO: Check size so Span can be reasonably constructed?
switch (aFormat) {
case AudioSampleFormat::U8:
case AudioSampleFormat::U8_planar:
return AsVariant(aSpan);
case AudioSampleFormat::S16:
case AudioSampleFormat::S16_planar:
return AsVariant(Span(reinterpret_cast<int16_t*>(aSpan.data()), Length));
case AudioSampleFormat::S32:
case AudioSampleFormat::S32_planar:
return AsVariant(Span(reinterpret_cast<int32_t*>(aSpan.data()), Length));
case AudioSampleFormat::F32:
case AudioSampleFormat::F32_planar:
return AsVariant(Span(reinterpret_cast<float*>(aSpan.data()), Length));
}
MOZ_ASSERT_UNREACHABLE("Invalid enum value");
return AsVariant(aSpan);
}
void CopySamples(DataSpanType& aSource, DataSpanType& aDest,
uint32_t aSourceChannelCount,
const AudioSampleFormat aSourceFormat,
const CopyToSpec& aCopyToSpec) {
aSource.match([&](auto& src) {
aDest.match([&](auto& dst) {
CopySamples(src, dst, aSourceChannelCount, aSourceFormat, aCopyToSpec);
});
});
}
void DoCopy(Span<uint8_t> aSource, Span<uint8_t> aDest,
const uint32_t aSourceChannelCount,
const AudioSampleFormat aSourceFormat,
const CopyToSpec& aCopyToSpec) {
DataSpanType source = GetDataSpan(aSource, aSourceFormat);
DataSpanType dest = GetDataSpan(aDest, aCopyToSpec.mFormat);
CopySamples(source, dest, aSourceChannelCount, aSourceFormat, aCopyToSpec);
}
// https://w3c.github.io/webcodecs/#dom-audiodata-copyto
void AudioData::CopyTo(const AllowSharedBufferSource& aDestination,
const AudioDataCopyToOptions& aOptions,
ErrorResult& aRv) {
AssertIsOnOwningThread();
size_t destLength = ProcessTypedArraysFixed(
aDestination, [&](const Span<uint8_t>& aData) -> size_t {
return aData.LengthBytes();
});
LOGD("AudioData::CopyTo {} -> {}", ToString().get(),
CopyToToString(destLength, aOptions).get(), 4);
if (!mResource) {
auto msg = "copyTo called on closed AudioData"_ns;
LOGD("{}", msg.get());
aRv.ThrowInvalidStateError(msg);
return;
}
uint64_t copyElementCount = ComputeCopyElementCount(aOptions, aRv);
if (aRv.Failed()) {
LOGD("AudioData::CopyTo failed in ComputeCopyElementCount");
return;
}
auto destFormat = mAudioSampleFormat;
if (aOptions.mFormat.WasPassed()) {
destFormat = OptionalToMaybe(aOptions.mFormat);
}
uint32_t bytesPerSample = BytesPerSamples(destFormat.value());
CheckedInt<uint32_t> copyLength = bytesPerSample;
copyLength *= copyElementCount;
if (copyLength.value() > destLength) {
auto msg = nsFmtCString(FMT_STRING("destination buffer of length {} too "
"small for copying {} elements"),
destLength, bytesPerSample * copyElementCount);
LOGD("{}", msg.get());
aRv.ThrowRangeError(msg);
return;
}
uint32_t framesToCopy = mNumberOfFrames - aOptions.mFrameOffset;
if (aOptions.mFrameCount.WasPassed()) {
framesToCopy = aOptions.mFrameCount.Value();
}
CopyToSpec copyToSpec(framesToCopy, aOptions.mFrameOffset,
aOptions.mPlaneIndex, destFormat.value());
// Now a couple layers of macros to type the pointers and perform the actual
// copy.
ProcessTypedArraysFixed(aDestination, [&](const Span<uint8_t>& aData) {
DoCopy(mResource->Data(), aData, mNumberOfChannels,
mAudioSampleFormat.value(), copyToSpec);
});
}
// https://w3c.github.io/webcodecs/#dom-audiodata-clone
already_AddRefed<AudioData> AudioData::Clone(ErrorResult& aRv) {
AssertIsOnOwningThread();
if (!mResource) {
auto msg = "No media resource in the AudioData now"_ns;
LOGD("{}", msg.get());
aRv.ThrowInvalidStateError(msg);
return nullptr;
}
return MakeAndAddRef<AudioData>(*this);
}
// https://w3c.github.io/webcodecs/#close-audiodata
void AudioData::Close() {
AssertIsOnOwningThread();
mResource = nullptr;
mSampleRate = 0;
mNumberOfFrames = 0;
mNumberOfChannels = 0;
mAudioSampleFormat = Nothing();
}
bool AudioData::IsClosed() const { return !mResource; }
// https://w3c.github.io/webcodecs/#ref-for-deserialization-steps%E2%91%A1
/* static */
JSObject* AudioData::ReadStructuredClone(JSContext* aCx,
nsIGlobalObject* aGlobal,
JSStructuredCloneReader* aReader,
const AudioDataSerializedData& aData) {
JS::Rooted<JS::Value> value(aCx, JS::NullValue());
// To avoid a rooting hazard error from returning a raw JSObject* before
// running the RefPtr destructor, RefPtr needs to be destructed before
// returning the raw JSObject*, which is why the RefPtr<AudioData> is created
// in the scope below. Otherwise, the static analysis infers the RefPtr cannot
// be safely destructed while the unrooted return JSObject* is on the stack.
{
RefPtr<AudioData> frame = MakeAndAddRef<AudioData>(aGlobal, aData);
if (!GetOrCreateDOMReflector(aCx, frame, &value) || !value.isObject()) {
LOGE("GetOrCreateDOMReflect failure");
return nullptr;
}
}
return value.toObjectOrNull();
}
// https://w3c.github.io/webcodecs/#ref-for-audiodata%E2%91%A2%E2%91%A2
bool AudioData::WriteStructuredClone(JSStructuredCloneWriter* aWriter,
StructuredCloneHolder* aHolder) const {
AssertIsOnOwningThread();
// AudioData closed
if (!mResource) {
LOGD("AudioData was already close in WriteStructuredClone");
return false;
}
const uint32_t index = aHolder->AudioData().Length();
// https://github.com/w3c/webcodecs/issues/717
// For now, serialization is only allowed in the same address space, it's OK
// to send a refptr here instead of copying the backing buffer.
aHolder->AudioData().AppendElement(AudioDataSerializedData(*this));
return !NS_WARN_IF(!JS_WriteUint32Pair(aWriter, SCTAG_DOM_AUDIODATA, index));
}
// https://w3c.github.io/webcodecs/#ref-for-transfer-steps
UniquePtr<AudioData::TransferredData> AudioData::Transfer() {
AssertIsOnOwningThread();
if (!mResource) {
// Closed
LOGD("AudioData was already close in Transfer");
return nullptr;
}
// This adds a ref to the resource
auto serialized = MakeUnique<AudioDataSerializedData>(*this);
// This removes the ref to the resource, effectively transfering the backing
// storage.
Close();
return serialized;
}
// https://w3c.github.io/webcodecs/#ref-for-transfer-receiving-steps
/* static */
already_AddRefed<AudioData> AudioData::FromTransferred(nsIGlobalObject* aGlobal,
TransferredData* aData) {
MOZ_ASSERT(aData);
return MakeAndAddRef<AudioData>(aGlobal, *aData);
}
void AudioData::CloseIfNeeded() {
if (mResource) {
mResource = nullptr;
}
}
RefPtr<mozilla::AudioData> AudioData::ToAudioData() const {
// Always convert to f32 interleaved for now, as this Gecko's prefered
// internal audio representation for encoding and decoding.
// mResource can be bigger than needed.
Span<uint8_t> data = mResource->Data();
CheckedUint64 sampleCount = mNumberOfFrames;
sampleCount *= mNumberOfChannels;
if (!sampleCount.isValid()) {
LOGE("Overflow AudioData::ToAudioData when computing the number of frames");
return nullptr;
}
AlignedAudioBuffer buf(sampleCount.value());
if (!buf.Length()) {
LOGE("OOM when allocating storage for AudioData conversion");
return nullptr;
}
Span<uint8_t> storage(reinterpret_cast<uint8_t*>(buf.Data()), buf.Size());
CopyToSpec spec(mNumberOfFrames, 0, 0, AudioSampleFormat::F32);
DoCopy(data, storage, mNumberOfChannels, mAudioSampleFormat.value(), spec);
return MakeRefPtr<mozilla::AudioData>(
0, media::TimeUnit::FromMicroseconds(mTimestamp), std::move(buf),
mNumberOfChannels, SaturatingCast<uint32_t>(mSampleRate));
}
#undef LOGD
#undef LOGE
} // namespace mozilla::dom
View git blame Copy permalink