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// Copyright 2020 The Chromium Authors. All rights reserved.
// 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/peerconnection/rtc_encoded_audio_underlying_sink.h"
#include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_encoded_audio_frame.h"
#include "third_party/blink/renderer/core/dom/dom_exception.h"
#include "third_party/blink/renderer/platform/bindings/exception_state.h"
#include "third_party/blink/renderer/platform/peerconnection/rtc_encoded_audio_stream_transformer.h"
#include "third_party/webrtc/api/frame_transformer_interface.h"
namespace blink {
RTCEncodedAudioUnderlyingSink::RTCEncodedAudioUnderlyingSink(
ScriptState* script_state,
TransformerCallback transformer_callback)
: transformer_callback_(std::move(transformer_callback)) {
DCHECK(transformer_callback_);
}
ScriptPromise RTCEncodedAudioUnderlyingSink::start(
ScriptState* script_state,
WritableStreamDefaultController* controller,
ExceptionState&) {
// No extra setup needed.
return ScriptPromise::CastUndefined(script_state);
}
ScriptPromise RTCEncodedAudioUnderlyingSink::write(
ScriptState* script_state,
ScriptValue chunk,
WritableStreamDefaultController* controller,
ExceptionState& exception_state) {
RTCEncodedAudioFrame* encoded_frame =
V8RTCEncodedAudioFrame::ToImplWithTypeCheck(script_state->GetIsolate(),
chunk.V8Value());
if (!encoded_frame) {
exception_state.ThrowTypeError("Invalid frame");
return ScriptPromise();
}
if (!transformer_callback_) {
exception_state.ThrowDOMException(DOMExceptionCode::kInvalidStateError,
"Stream closed");
return ScriptPromise();
}
auto webrtc_frame = encoded_frame->PassWebRtcFrame();
if (!webrtc_frame) {
exception_state.ThrowDOMException(DOMExceptionCode::kOperationError,
"Empty frame");
return ScriptPromise();
}
RTCEncodedAudioStreamTransformer* transformer = transformer_callback_.Run();
if (!transformer) {
exception_state.ThrowDOMException(DOMExceptionCode::kInvalidStateError,
"No underlying sink");
return ScriptPromise();
}
transformer->SendFrameToSink(std::move(webrtc_frame));
return ScriptPromise::CastUndefined(script_state);
}
ScriptPromise RTCEncodedAudioUnderlyingSink::close(ScriptState* script_state,
ExceptionState&) {
// Disconnect from the transformer if the sink is closed.
if (transformer_callback_)
transformer_callback_.Reset();
return ScriptPromise::CastUndefined(script_state);
}
ScriptPromise RTCEncodedAudioUnderlyingSink::abort(
ScriptState* script_state,
ScriptValue reason,
ExceptionState& exception_state) {
// It is not possible to cancel any frames already sent to the WebRTC sink,
// thus abort() has the same effect as close().
return close(script_state, exception_state);
}
void RTCEncodedAudioUnderlyingSink::Trace(Visitor* visitor) {
UnderlyingSinkBase::Trace(visitor);
}
} // namespace blink
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