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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_source.h"
#include "third_party/blink/renderer/bindings/core/v8/v8_throw_dom_exception.h"
#include "third_party/blink/renderer/core/streams/readable_stream_default_controller_with_script_scope.h"
#include "third_party/blink/renderer/modules/peerconnection/rtc_encoded_audio_frame.h"
#include "third_party/blink/renderer/platform/bindings/exception_code.h"
#include "third_party/blink/renderer/platform/bindings/exception_state.h"
#include "third_party/blink/renderer/platform/wtf/vector.h"
#include "third_party/webrtc/api/frame_transformer_interface.h"
namespace blink {
// Frames should not be queued at all. We allow queuing a few frames to deal
// with transient slowdowns. Specified as a negative number of frames since
// queuing is reported by the stream controller as a negative desired size.
const int RTCEncodedAudioUnderlyingSource::kMinQueueDesiredSize = -60;
RTCEncodedAudioUnderlyingSource::RTCEncodedAudioUnderlyingSource(
ScriptState* script_state,
base::OnceClosure disconnect_callback,
bool is_receiver)
: UnderlyingSourceBase(script_state),
script_state_(script_state),
disconnect_callback_(std::move(disconnect_callback)),
is_receiver_(is_receiver) {
DCHECK(disconnect_callback_);
}
ScriptPromise RTCEncodedAudioUnderlyingSource::pull(ScriptState* script_state) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
// WebRTC is a push source without backpressure support, so nothing to do
// here.
return ScriptPromise::CastUndefined(script_state);
}
ScriptPromise RTCEncodedAudioUnderlyingSource::Cancel(ScriptState* script_state,
ScriptValue reason) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (disconnect_callback_)
std::move(disconnect_callback_).Run();
return ScriptPromise::CastUndefined(script_state);
}
void RTCEncodedAudioUnderlyingSource::Trace(Visitor* visitor) const {
visitor->Trace(script_state_);
UnderlyingSourceBase::Trace(visitor);
}
void RTCEncodedAudioUnderlyingSource::OnFrameFromSource(
std::unique_ptr<webrtc::TransformableFrameInterface> webrtc_frame) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
// If the source is canceled or there are too many queued frames,
// drop the new frame.
if (!disconnect_callback_ || !Controller() ||
Controller()->DesiredSize() <= kMinQueueDesiredSize) {
return;
}
RTCEncodedAudioFrame* encoded_frame = nullptr;
if (is_receiver_) {
// Receivers produce frames as webrtc::TransformableAudioFrameInterface,
// which allows exposing the CSRCs.
std::unique_ptr<webrtc::TransformableAudioFrameInterface> audio_frame =
base::WrapUnique(static_cast<webrtc::TransformableAudioFrameInterface*>(
webrtc_frame.release()));
encoded_frame =
MakeGarbageCollected<RTCEncodedAudioFrame>(std::move(audio_frame));
} else {
encoded_frame =
MakeGarbageCollected<RTCEncodedAudioFrame>(std::move(webrtc_frame));
}
Controller()->Enqueue(encoded_frame);
}
void RTCEncodedAudioUnderlyingSource::Close() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (disconnect_callback_)
std::move(disconnect_callback_).Run();
if (Controller())
Controller()->Close();
}
} // namespace blink
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