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// Copyright 2020 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/peerconnection/rtc_encoded_audio_underlying_source.h"
#include "base/memory/ptr_util.h"
#include "base/unguessable_token.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/modules/peerconnection/rtc_encoded_audio_frame_delegate.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/heap/cross_thread_persistent.h"
#include "third_party/blink/renderer/platform/scheduler/public/post_cross_thread_task.h"
#include "third_party/blink/renderer/platform/wtf/cross_thread_functional.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,
WTF::CrossThreadOnceClosure disconnect_callback)
: blink::RTCEncodedAudioUnderlyingSource(
script_state,
std::move(disconnect_callback),
/*enable_frame_restrictions=*/false,
base::UnguessableToken::Null(),
/*controller_override=*/nullptr) {}
RTCEncodedAudioUnderlyingSource::RTCEncodedAudioUnderlyingSource(
ScriptState* script_state,
WTF::CrossThreadOnceClosure disconnect_callback,
bool enable_frame_restrictions,
base::UnguessableToken owner_id,
ReadableStreamDefaultControllerWithScriptScope* override_controller)
: UnderlyingSourceBase(script_state),
script_state_(script_state),
disconnect_callback_(std::move(disconnect_callback)),
override_controller_(override_controller),
enable_frame_restrictions_(enable_frame_restrictions),
owner_id_(owner_id) {
DCHECK(disconnect_callback_);
ExecutionContext* context = ExecutionContext::From(script_state);
task_runner_ = context->GetTaskRunner(TaskType::kInternalMediaRealTime);
}
ReadableStreamDefaultControllerWithScriptScope*
RTCEncodedAudioUnderlyingSource::GetController() {
if (override_controller_) {
return override_controller_;
}
return Controller();
}
ScriptPromise<IDLUndefined> RTCEncodedAudioUnderlyingSource::Pull(
ScriptState* script_state,
ExceptionState&) {
DCHECK(task_runner_->BelongsToCurrentThread());
// WebRTC is a push source without backpressure support, so nothing to do
// here.
return ToResolvedUndefinedPromise(script_state);
}
ScriptPromise<IDLUndefined> RTCEncodedAudioUnderlyingSource::Cancel(
ScriptState* script_state,
ScriptValue reason,
ExceptionState&) {
DCHECK(task_runner_->BelongsToCurrentThread());
if (disconnect_callback_)
std::move(disconnect_callback_).Run();
return ToResolvedUndefinedPromise(script_state);
}
void RTCEncodedAudioUnderlyingSource::Trace(Visitor* visitor) const {
visitor->Trace(script_state_);
visitor->Trace(override_controller_);
UnderlyingSourceBase::Trace(visitor);
}
void RTCEncodedAudioUnderlyingSource::OnFrameFromSource(
std::unique_ptr<webrtc::TransformableAudioFrameInterface> webrtc_frame) {
// It can happen that a frame is posted to the task runner of the old
// execution context during a stream transfer to a new context.
// TODO(https://crbug.com/1506631): Make the state updates related to the
// transfer atomic and turn this into a DCHECK.
if (!task_runner_->BelongsToCurrentThread()) {
DVLOG(1) << "Dropped frame posted to incorrect task runner. This can "
"happen during transfer.";
return;
}
// If the source is canceled or there are too many queued frames,
// drop the new frame.
if (!disconnect_callback_ || !GetExecutionContext()) {
return;
}
if (!GetController()) {
// TODO(ricea): Maybe avoid dropping frames during transfer?
DVLOG(1) << "Dropped frame due to null Controller(). This can happen "
"during transfer.";
return;
}
if (GetController()->DesiredSize() <= kMinQueueDesiredSize) {
dropped_frames_++;
VLOG_IF(2, (dropped_frames_ % 20 == 0))
<< "Dropped total of " << dropped_frames_
<< " encoded audio frames due to too many already being queued.";
return;
}
RTCEncodedAudioFrame* encoded_frame;
if (enable_frame_restrictions_) {
encoded_frame = MakeGarbageCollected<RTCEncodedAudioFrame>(
std::move(webrtc_frame), owner_id_, ++last_enqueued_frame_counter_);
} else {
encoded_frame =
MakeGarbageCollected<RTCEncodedAudioFrame>(std::move(webrtc_frame));
}
GetController()->Enqueue(encoded_frame);
}
void RTCEncodedAudioUnderlyingSource::Close() {
DCHECK(task_runner_->BelongsToCurrentThread());
if (disconnect_callback_)
std::move(disconnect_callback_).Run();
if (GetController()) {
GetController()->Close();
}
}
void RTCEncodedAudioUnderlyingSource::OnSourceTransferStartedOnTaskRunner() {
DCHECK(task_runner_->BelongsToCurrentThread());
// This can potentially be called before the stream is constructed and so
// Controller() is still unset.
if (GetController()) {
GetController()->Close();
}
}
void RTCEncodedAudioUnderlyingSource::OnSourceTransferStarted() {
PostCrossThreadTask(
*task_runner_, FROM_HERE,
CrossThreadBindOnce(
&RTCEncodedAudioUnderlyingSource::OnSourceTransferStartedOnTaskRunner,
WrapCrossThreadPersistent(this)));
}
} // namespace blink
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