| // Copyright 2021 The Chromium Authors |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "third_party/blink/renderer/modules/webcodecs/audio_encoder.h" |
| |
| #include <cinttypes> |
| #include <limits> |
| |
| #include "base/containers/contains.h" |
| #include "base/feature_list.h" |
| #include "base/metrics/histogram_functions.h" |
| #include "base/numerics/safe_conversions.h" |
| #include "base/trace_event/common/trace_event_common.h" |
| #include "base/trace_event/trace_event.h" |
| #include "build/build_config.h" |
| #include "media/audio/audio_opus_encoder.h" |
| #include "media/base/audio_parameters.h" |
| #include "media/base/limits.h" |
| #include "media/base/media_switches.h" |
| #include "media/base/mime_util.h" |
| #include "media/base/offloading_audio_encoder.h" |
| #include "media/mojo/clients/mojo_audio_encoder.h" |
| #include "media/mojo/mojom/audio_encoder.mojom-blink.h" |
| #include "media/mojo/mojom/interface_factory.mojom.h" |
| #include "third_party/blink/public/common/thread_safe_browser_interface_broker_proxy.h" |
| #include "third_party/blink/public/mojom/use_counter/metrics/web_feature.mojom-blink.h" |
| #include "third_party/blink/public/platform/platform.h" |
| #include "third_party/blink/renderer/bindings/core/v8/to_v8_traits.h" |
| #include "third_party/blink/renderer/bindings/core/v8/v8_typedefs.h" |
| #include "third_party/blink/renderer/bindings/modules/v8/v8_aac_encoder_config.h" |
| #include "third_party/blink/renderer/bindings/modules/v8/v8_audio_data_init.h" |
| #include "third_party/blink/renderer/bindings/modules/v8/v8_audio_decoder_config.h" |
| #include "third_party/blink/renderer/bindings/modules/v8/v8_audio_encoder_config.h" |
| #include "third_party/blink/renderer/bindings/modules/v8/v8_audio_encoder_support.h" |
| #include "third_party/blink/renderer/bindings/modules/v8/v8_encoded_audio_chunk_metadata.h" |
| #include "third_party/blink/renderer/bindings/modules/v8/v8_opus_encoder_config.h" |
| #include "third_party/blink/renderer/modules/webaudio/audio_buffer.h" |
| #include "third_party/blink/renderer/modules/webcodecs/allow_shared_buffer_source_util.h" |
| #include "third_party/blink/renderer/modules/webcodecs/encoded_audio_chunk.h" |
| #include "third_party/blink/renderer/platform/bindings/exception_state.h" |
| #include "third_party/blink/renderer/platform/heap/cross_thread_handle.h" |
| #include "third_party/blink/renderer/platform/wtf/cross_thread_functional.h" |
| #include "third_party/blink/renderer/platform/wtf/text/string_builder.h" |
| |
| namespace blink { |
| |
| namespace { |
| |
| constexpr const char kCategory[] = "media"; |
| |
| #if BUILDFLAG(IS_ANDROID) || BUILDFLAG(IS_IOS) |
| constexpr uint32_t kDefaultOpusComplexity = 5; |
| #else |
| constexpr uint32_t kDefaultOpusComplexity = 9; |
| #endif |
| |
| template <typename T> |
| bool VerifyParameterValues(const T& value, |
| ExceptionState* exception_state, |
| WTF::String error_message, |
| WTF::Vector<T> supported_values) { |
| if (base::Contains(supported_values, value)) |
| return true; |
| |
| if (exception_state) { |
| WTF::StringBuilder error_builder; |
| error_builder.Append(error_message); |
| error_builder.Append(" Supported values: "); |
| for (auto i = 0u; i < supported_values.size(); i++) { |
| if (i != 0) |
| error_builder.Append(", "); |
| error_builder.AppendNumber(supported_values[i]); |
| } |
| exception_state->ThrowDOMException(DOMExceptionCode::kNotSupportedError, |
| error_builder.ToString()); |
| } |
| return false; |
| } |
| |
| AudioEncoderTraits::ParsedConfig* ParseAacConfigStatic( |
| const AacEncoderConfig* aac_config, |
| AudioEncoderTraits::ParsedConfig* result, |
| ExceptionState& exception_state) { |
| result->options.aac = media::AudioEncoder::AacOptions(); |
| switch (aac_config->format().AsEnum()) { |
| case V8AacBitstreamFormat::Enum::kAac: |
| result->options.aac->format = media::AudioEncoder::AacOutputFormat::AAC; |
| return result; |
| case V8AacBitstreamFormat::Enum::kAdts: |
| result->options.aac->format = media::AudioEncoder::AacOutputFormat::ADTS; |
| return result; |
| } |
| return result; |
| } |
| |
| AudioEncoderTraits::ParsedConfig* ParseOpusConfigStatic( |
| const OpusEncoderConfig* opus_config, |
| AudioEncoderTraits::ParsedConfig* result, |
| ExceptionState& exception_state) { |
| constexpr uint32_t kComplexityUpperBound = 10; |
| uint32_t complexity = opus_config->getComplexityOr(kDefaultOpusComplexity); |
| if (complexity > kComplexityUpperBound) { |
| exception_state.ThrowTypeError( |
| ExceptionMessages::IndexExceedsMaximumBound<uint32_t>( |
| "Opus complexity", complexity, kComplexityUpperBound)); |
| return nullptr; |
| } |
| |
| constexpr uint32_t kPacketLossPercUpperBound = 100; |
| uint32_t packet_loss_perc = opus_config->packetlossperc(); |
| if (packet_loss_perc > kPacketLossPercUpperBound) { |
| exception_state.ThrowTypeError( |
| ExceptionMessages::IndexExceedsMaximumBound<uint32_t>( |
| "Opus packetlossperc", packet_loss_perc, |
| kPacketLossPercUpperBound)); |
| return nullptr; |
| } |
| |
| // `frame_duration` must be a valid frame duration, defined in section 2.1.4. |
| // of RFC6716. |
| constexpr base::TimeDelta kFrameDurationLowerBound = base::Microseconds(2500); |
| constexpr base::TimeDelta kFrameDurationUpperBound = base::Milliseconds(120); |
| uint64_t frame_duration = opus_config->frameDuration(); |
| if (frame_duration < kFrameDurationLowerBound.InMicroseconds() || |
| frame_duration > kFrameDurationUpperBound.InMicroseconds()) { |
| exception_state.ThrowTypeError( |
| ExceptionMessages::IndexOutsideRange<uint64_t>( |
| "Opus frameDuration", frame_duration, |
| kFrameDurationLowerBound.InMicroseconds(), |
| ExceptionMessages::BoundType::kInclusiveBound, |
| kFrameDurationUpperBound.InMicroseconds(), |
| ExceptionMessages::BoundType::kInclusiveBound)); |
| return nullptr; |
| } |
| |
| // Any multiple of a frame duration is allowed by RFC6716. Concretely, this |
| // means any multiple of 2500 microseconds. |
| if (frame_duration % kFrameDurationLowerBound.InMicroseconds() != 0) { |
| exception_state.ThrowTypeError(String::Format( |
| "Invalid Opus frameDuration; expected a multiple of %" PRIu64 |
| ", received %" PRIu64 ".", |
| kFrameDurationLowerBound.InMicroseconds(), frame_duration)); |
| return nullptr; |
| } |
| |
| // TODO(crbug.com/1378399): Support all multiples of basic frame durations. |
| if (!VerifyParameterValues(frame_duration, &exception_state, |
| "Unsupported Opus frameDuration.", |
| {2500, 5000, 10000, 20000, 40000, 60000})) { |
| return nullptr; |
| } |
| |
| if (opus_config->format().AsEnum() == V8OpusBitstreamFormat::Enum::kOgg) { |
| exception_state.ThrowDOMException(DOMExceptionCode::kNotSupportedError, |
| "Opus Ogg format is unsupported"); |
| return nullptr; |
| } |
| |
| result->options.opus = { |
| .frame_duration = base::Microseconds(frame_duration), |
| .complexity = complexity, |
| .packet_loss_perc = packet_loss_perc, |
| .use_in_band_fec = opus_config->useinbandfec(), |
| .use_dtx = opus_config->usedtx(), |
| }; |
| |
| return result; |
| } |
| |
| AudioEncoderTraits::ParsedConfig* ParseConfigStatic( |
| const AudioEncoderConfig* config, |
| ExceptionState& exception_state) { |
| if (!config) { |
| exception_state.ThrowTypeError("No config provided"); |
| return nullptr; |
| } |
| auto* result = MakeGarbageCollected<AudioEncoderTraits::ParsedConfig>(); |
| |
| result->options.codec = media::AudioCodec::kUnknown; |
| bool is_codec_ambiguous = true; |
| bool parse_succeeded = ParseAudioCodecString( |
| "", config->codec().Utf8(), &is_codec_ambiguous, &result->options.codec); |
| |
| if (!parse_succeeded || is_codec_ambiguous) { |
| exception_state.ThrowTypeError("Unknown codec."); |
| return nullptr; |
| } |
| |
| result->options.channels = config->numberOfChannels(); |
| if (result->options.channels < 1 || |
| result->options.channels > media::limits::kMaxChannels) { |
| exception_state.ThrowTypeError(String::Format( |
| "Invalid channel number; expected range from %d to %d, received %d.", 1, |
| media::limits::kMaxChannels, result->options.channels)); |
| return nullptr; |
| } |
| |
| result->options.sample_rate = config->sampleRate(); |
| if (result->options.sample_rate < media::limits::kMinSampleRate || |
| result->options.sample_rate > media::limits::kMaxSampleRate) { |
| exception_state.ThrowTypeError(String::Format( |
| "Invalid sample rate; expected range from %d to %d, received %d.", |
| media::limits::kMinSampleRate, media::limits::kMaxSampleRate, |
| result->options.sample_rate)); |
| return nullptr; |
| } |
| |
| result->codec_string = config->codec(); |
| if (config->hasBitrate()) { |
| if (config->bitrate() > std::numeric_limits<int>::max()) { |
| exception_state.ThrowTypeError(String::Format( |
| "Bitrate is too large; expected at most %d, received %" PRIu64, |
| std::numeric_limits<int>::max(), config->bitrate())); |
| return nullptr; |
| } |
| result->options.bitrate = static_cast<int>(config->bitrate()); |
| } |
| |
| switch (result->options.codec) { |
| case media::AudioCodec::kOpus: |
| return ParseOpusConfigStatic( |
| config->hasOpus() ? config->opus() : OpusEncoderConfig::Create(), |
| result, exception_state); |
| case media::AudioCodec::kAAC: { |
| auto* aac_config = |
| config->hasAac() ? config->aac() : AacEncoderConfig::Create(); |
| return ParseAacConfigStatic(aac_config, result, exception_state); |
| } |
| default: |
| return result; |
| } |
| } |
| |
| bool VerifyCodecSupportStatic(AudioEncoderTraits::ParsedConfig* config, |
| ExceptionState* exception_state) { |
| switch (config->options.codec) { |
| case media::AudioCodec::kOpus: { |
| if (config->options.channels > 2) { |
| // Our Opus implementation only supports up to 2 channels |
| if (exception_state) { |
| exception_state->ThrowDOMException( |
| DOMExceptionCode::kNotSupportedError, |
| String::Format("Too many channels for Opus encoder; " |
| "expected at most 2, received %d.", |
| config->options.channels)); |
| } |
| return false; |
| } |
| if (config->options.bitrate.has_value() && |
| config->options.bitrate.value() < |
| media::AudioOpusEncoder::kMinBitrate) { |
| if (exception_state) { |
| exception_state->ThrowDOMException( |
| DOMExceptionCode::kNotSupportedError, |
| String::Format( |
| "Opus bitrate is too low; expected at least %d, received %d.", |
| media::AudioOpusEncoder::kMinBitrate, |
| config->options.bitrate.value())); |
| } |
| return false; |
| } |
| return true; |
| } |
| case media::AudioCodec::kAAC: { |
| if (base::FeatureList::IsEnabled(media::kPlatformAudioEncoder)) { |
| if (!VerifyParameterValues(config->options.channels, exception_state, |
| "Unsupported number of channels.", |
| {1, 2, 6})) { |
| return false; |
| } |
| if (config->options.bitrate.has_value()) { |
| if (!VerifyParameterValues(config->options.bitrate.value(), |
| exception_state, "Unsupported bitrate.", |
| {96000, 128000, 160000, 192000})) { |
| return false; |
| } |
| } |
| if (!VerifyParameterValues(config->options.sample_rate, exception_state, |
| "Unsupported sample rate.", |
| {44100, 48000})) { |
| return false; |
| } |
| return true; |
| } |
| [[fallthrough]]; |
| } |
| default: |
| if (exception_state) { |
| exception_state->ThrowDOMException(DOMExceptionCode::kNotSupportedError, |
| "Unsupported codec type."); |
| } |
| return false; |
| } |
| } |
| |
| AacEncoderConfig* CopyAacConfig(const AacEncoderConfig& config) { |
| auto* result = AacEncoderConfig::Create(); |
| result->setFormat(config.format()); |
| return result; |
| } |
| |
| OpusEncoderConfig* CopyOpusConfig(const OpusEncoderConfig& config) { |
| auto* opus_result = OpusEncoderConfig::Create(); |
| opus_result->setFormat(config.format()); |
| opus_result->setFrameDuration(config.frameDuration()); |
| opus_result->setComplexity(config.getComplexityOr(kDefaultOpusComplexity)); |
| opus_result->setPacketlossperc(config.packetlossperc()); |
| opus_result->setUseinbandfec(config.useinbandfec()); |
| opus_result->setUsedtx(config.usedtx()); |
| return opus_result; |
| } |
| |
| AudioEncoderConfig* CopyConfig(const AudioEncoderConfig& config) { |
| auto* result = AudioEncoderConfig::Create(); |
| result->setCodec(config.codec()); |
| result->setSampleRate(config.sampleRate()); |
| result->setNumberOfChannels(config.numberOfChannels()); |
| if (config.hasBitrate()) |
| result->setBitrate(config.bitrate()); |
| |
| if (config.codec() == String("opus") && config.hasOpus()) |
| result->setOpus(CopyOpusConfig(*config.opus())); |
| if (config.codec() == String("aac") && config.hasAac()) { |
| result->setAac(CopyAacConfig(*config.aac())); |
| } |
| |
| return result; |
| } |
| |
| std::unique_ptr<media::AudioEncoder> CreateSoftwareAudioEncoder( |
| media::AudioCodec codec) { |
| if (codec != media::AudioCodec::kOpus) |
| return nullptr; |
| auto software_encoder = std::make_unique<media::AudioOpusEncoder>(); |
| return std::make_unique<media::OffloadingAudioEncoder>( |
| std::move(software_encoder)); |
| } |
| |
| std::unique_ptr<media::AudioEncoder> CreatePlatformAudioEncoder( |
| media::AudioCodec codec) { |
| if (codec != media::AudioCodec::kAAC) |
| return nullptr; |
| |
| mojo::PendingRemote<media::mojom::InterfaceFactory> pending_interface_factory; |
| mojo::Remote<media::mojom::InterfaceFactory> interface_factory; |
| Platform::Current()->GetBrowserInterfaceBroker()->GetInterface( |
| pending_interface_factory.InitWithNewPipeAndPassReceiver()); |
| interface_factory.Bind(std::move(pending_interface_factory)); |
| |
| mojo::PendingRemote<media::mojom::AudioEncoder> encoder_remote; |
| interface_factory->CreateAudioEncoder( |
| encoder_remote.InitWithNewPipeAndPassReceiver()); |
| return std::make_unique<media::MojoAudioEncoder>(std::move(encoder_remote)); |
| } |
| |
| } // namespace |
| |
| // static |
| const char* AudioEncoderTraits::GetName() { |
| return "AudioEncoder"; |
| } |
| |
| AudioEncoder* AudioEncoder::Create(ScriptState* script_state, |
| const AudioEncoderInit* init, |
| ExceptionState& exception_state) { |
| auto* result = |
| MakeGarbageCollected<AudioEncoder>(script_state, init, exception_state); |
| return exception_state.HadException() ? nullptr : result; |
| } |
| |
| AudioEncoder::AudioEncoder(ScriptState* script_state, |
| const AudioEncoderInit* init, |
| ExceptionState& exception_state) |
| : Base(script_state, init, exception_state) { |
| UseCounter::Count(ExecutionContext::From(script_state), |
| WebFeature::kWebCodecs); |
| } |
| |
| AudioEncoder::~AudioEncoder() = default; |
| |
| std::unique_ptr<media::AudioEncoder> AudioEncoder::CreateMediaAudioEncoder( |
| const ParsedConfig& config) { |
| if (auto result = CreatePlatformAudioEncoder(config.options.codec)) |
| return result; |
| return CreateSoftwareAudioEncoder(config.options.codec); |
| } |
| |
| void AudioEncoder::ProcessConfigure(Request* request) { |
| DCHECK_NE(state_.AsEnum(), V8CodecState::Enum::kClosed); |
| DCHECK_EQ(request->type, Request::Type::kConfigure); |
| DCHECK(active_config_); |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| |
| request->StartTracing(); |
| |
| media_encoder_ = CreateMediaAudioEncoder(*active_config_); |
| if (!media_encoder_) { |
| HandleError(logger_->MakeOperationError( |
| "Encoder creation error.", |
| media::EncoderStatus( |
| media::EncoderStatus::Codes::kEncoderInitializationError, |
| "Unable to create encoder (most likely unsupported " |
| "codec/acceleration requirement combination)"))); |
| request->EndTracing(); |
| return; |
| } |
| |
| auto output_cb = ConvertToBaseRepeatingCallback(CrossThreadBindRepeating( |
| &AudioEncoder::CallOutputCallback, |
| MakeUnwrappingCrossThreadWeakHandle(this), |
| // We can't use |active_config_| from |this| because it can change by |
| // the time the callback is executed. |
| MakeUnwrappingCrossThreadHandle(active_config_.Get()), reset_count_)); |
| |
| auto done_callback = [](AudioEncoder* self, media::AudioCodec codec, |
| Request* req, media::EncoderStatus status) { |
| if (!self || self->reset_count_ != req->reset_count) { |
| req->EndTracing(/*aborted=*/true); |
| return; |
| } |
| DCHECK_CALLED_ON_VALID_SEQUENCE(self->sequence_checker_); |
| if (!status.is_ok()) { |
| self->HandleError(self->logger_->MakeOperationError("Encoding error.", |
| std::move(status))); |
| } else { |
| base::UmaHistogramEnumeration("Blink.WebCodecs.AudioEncoder.Codec", |
| codec); |
| } |
| |
| req->EndTracing(); |
| self->blocking_request_in_progress_ = false; |
| self->ProcessRequests(); |
| }; |
| |
| blocking_request_in_progress_ = true; |
| first_output_after_configure_ = true; |
| media_encoder_->Initialize( |
| active_config_->options, std::move(output_cb), |
| ConvertToBaseOnceCallback(CrossThreadBindOnce( |
| done_callback, MakeUnwrappingCrossThreadWeakHandle(this), |
| active_config_->options.codec, |
| MakeUnwrappingCrossThreadHandle(request)))); |
| } |
| |
| void AudioEncoder::ProcessEncode(Request* request) { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| DCHECK_EQ(state_, V8CodecState::Enum::kConfigured); |
| DCHECK(media_encoder_); |
| DCHECK_EQ(request->type, Request::Type::kEncode); |
| DCHECK_GT(requested_encodes_, 0u); |
| |
| request->StartTracing(); |
| |
| auto* audio_data = request->input.Release(); |
| |
| auto data = audio_data->data(); |
| |
| // The data shouldn't be closed at this point. |
| DCHECK(data); |
| |
| auto done_callback = [](AudioEncoder* self, Request* req, |
| media::EncoderStatus status) { |
| if (!self || self->reset_count_ != req->reset_count) { |
| req->EndTracing(/*aborted=*/true); |
| return; |
| } |
| DCHECK_CALLED_ON_VALID_SEQUENCE(self->sequence_checker_); |
| if (!status.is_ok()) { |
| self->HandleError(self->logger_->MakeEncodingError("Encoding error.", |
| std::move(status))); |
| } |
| |
| req->EndTracing(); |
| self->ProcessRequests(); |
| }; |
| |
| if (data->channel_count() != active_config_->options.channels || |
| data->sample_rate() != active_config_->options.sample_rate) { |
| HandleError(logger_->MakeEncodingError( |
| "Input audio buffer is incompatible with codec parameters", |
| media::EncoderStatus(media::EncoderStatus::Codes::kEncoderFailedEncode) |
| .WithData("channels", data->channel_count()) |
| .WithData("sampleRate", data->sample_rate()))); |
| |
| request->EndTracing(); |
| |
| audio_data->close(); |
| return; |
| } |
| |
| // If |data|'s memory layout allows it, |audio_bus| will be a simple wrapper |
| // around it. Otherwise, |audio_bus| will contain a converted copy of |data|. |
| auto audio_bus = media::AudioBuffer::WrapOrCopyToAudioBus(data); |
| |
| base::TimeTicks timestamp = base::TimeTicks() + data->timestamp(); |
| |
| --requested_encodes_; |
| ScheduleDequeueEvent(); |
| media_encoder_->Encode( |
| std::move(audio_bus), timestamp, |
| ConvertToBaseOnceCallback(CrossThreadBindOnce( |
| done_callback, MakeUnwrappingCrossThreadWeakHandle(this), |
| MakeUnwrappingCrossThreadHandle(request)))); |
| |
| audio_data->close(); |
| } |
| |
| void AudioEncoder::ProcessReconfigure(Request* request) { |
| // Audio decoders don't currently support any meaningful reconfiguring |
| } |
| |
| AudioEncoder::ParsedConfig* AudioEncoder::ParseConfig( |
| const AudioEncoderConfig* opts, |
| ExceptionState& exception_state) { |
| return ParseConfigStatic(opts, exception_state); |
| } |
| |
| bool AudioEncoder::CanReconfigure(ParsedConfig& original_config, |
| ParsedConfig& new_config) { |
| return original_config.options.codec == new_config.options.codec && |
| original_config.options.channels == new_config.options.channels && |
| original_config.options.bitrate == new_config.options.bitrate && |
| original_config.options.sample_rate == new_config.options.sample_rate; |
| } |
| |
| bool AudioEncoder::VerifyCodecSupport(ParsedConfig* config, |
| ExceptionState& exception_state) { |
| return VerifyCodecSupportStatic(config, &exception_state); |
| } |
| |
| void AudioEncoder::CallOutputCallback( |
| ParsedConfig* active_config, |
| uint32_t reset_count, |
| media::EncodedAudioBuffer encoded_buffer, |
| absl::optional<media::AudioEncoder::CodecDescription> codec_desc) { |
| DCHECK(active_config); |
| if (!script_state_->ContextIsValid() || !output_callback_ || |
| state_.AsEnum() != V8CodecState::Enum::kConfigured || |
| reset_count != reset_count_) { |
| return; |
| } |
| |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| |
| MarkCodecActive(); |
| |
| auto buffer = media::DecoderBuffer::FromArray( |
| std::move(encoded_buffer.encoded_data), encoded_buffer.encoded_data_size); |
| buffer->set_timestamp(encoded_buffer.timestamp - base::TimeTicks()); |
| buffer->set_is_key_frame(true); |
| buffer->set_duration(encoded_buffer.duration); |
| auto* chunk = MakeGarbageCollected<EncodedAudioChunk>(std::move(buffer)); |
| |
| auto* metadata = MakeGarbageCollected<EncodedAudioChunkMetadata>(); |
| if (first_output_after_configure_ || codec_desc.has_value()) { |
| first_output_after_configure_ = false; |
| auto* decoder_config = MakeGarbageCollected<AudioDecoderConfig>(); |
| decoder_config->setCodec(active_config->codec_string); |
| decoder_config->setSampleRate(encoded_buffer.params.sample_rate()); |
| decoder_config->setNumberOfChannels(active_config->options.channels); |
| if (codec_desc.has_value()) { |
| auto* desc_array_buf = DOMArrayBuffer::Create(codec_desc.value().data(), |
| codec_desc.value().size()); |
| decoder_config->setDescription( |
| MakeGarbageCollected<AllowSharedBufferSource>(desc_array_buf)); |
| } |
| metadata->setDecoderConfig(decoder_config); |
| } |
| |
| TRACE_EVENT_BEGIN1(kCategory, GetTraceNames()->output.c_str(), "timestamp", |
| chunk->timestamp()); |
| |
| ScriptState::Scope scope(script_state_); |
| output_callback_->InvokeAndReportException(nullptr, chunk, metadata); |
| |
| TRACE_EVENT_END0(kCategory, GetTraceNames()->output.c_str()); |
| } |
| |
| // static |
| ScriptPromise AudioEncoder::isConfigSupported(ScriptState* script_state, |
| const AudioEncoderConfig* config, |
| ExceptionState& exception_state) { |
| auto* parsed_config = ParseConfigStatic(config, exception_state); |
| if (!parsed_config) { |
| DCHECK(exception_state.HadException()); |
| return ScriptPromise(); |
| } |
| |
| auto* support = AudioEncoderSupport::Create(); |
| support->setSupported(VerifyCodecSupportStatic(parsed_config, nullptr)); |
| support->setConfig(CopyConfig(*config)); |
| |
| return ScriptPromise::Cast( |
| script_state, ToV8Traits<AudioEncoderSupport>::ToV8(script_state, support) |
| .ToLocalChecked()); |
| } |
| |
| const AtomicString& AudioEncoder::InterfaceName() const { |
| return event_target_names::kAudioEncoder; |
| } |
| |
| } // namespace blink |
