blob: 30e63ff98bd431151906231a97a8ce360668e881 [file] [log] [blame]
* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <memory>
#include <string>
#include "api/frame_transformer_interface.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "rtc_base/buffer.h"
#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
// Delegates calls to FrameTransformerInterface to transform frames, and to
// ChannelSend to send the transformed frames using `send_frame_callback_` on
// the `encoder_queue_`.
// OnTransformedFrame() can be called from any thread, the delegate ensures
// thread-safe access to the ChannelSend callback.
class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback {
using SendFrameCallback =
std::function<int32_t(AudioFrameType frameType,
uint8_t payloadType,
uint32_t rtp_timestamp_with_offset,
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs)>;
SendFrameCallback send_frame_callback,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
TaskQueueBase* encoder_queue);
// Registers `this` as callback for `frame_transformer_`, to get the
// transformed frames.
void Init();
// Unregisters and releases the `frame_transformer_` reference, and resets
// `send_frame_callback_` under lock. Called from ChannelSend destructor to
// prevent running the callback on a dangling channel.
void Reset();
// Delegates the call to FrameTransformerInterface::TransformFrame, to
// transform the frame asynchronously.
void Transform(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t rtp_timestamp,
const uint8_t* payload_data,
size_t payload_size,
int64_t absolute_capture_timestamp_ms,
uint32_t ssrc,
const std::string& codec_mime_type);
// Implements TransformedFrameCallback. Can be called on any thread.
void OnTransformedFrame(
std::unique_ptr<TransformableFrameInterface> frame) override;
void StartShortCircuiting() override;
// Delegates the call to ChannelSend::SendRtpAudio on the `encoder_queue_`,
// by calling `send_audio_callback_`.
void SendFrame(std::unique_ptr<TransformableFrameInterface> frame) const;
~ChannelSendFrameTransformerDelegate() override = default;
mutable Mutex send_lock_;
SendFrameCallback send_frame_callback_ RTC_GUARDED_BY(send_lock_);
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_;
TaskQueueBase* const encoder_queue_;
bool short_circuit_ RTC_GUARDED_BY(send_lock_) = false;
std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
TransformableAudioFrameInterface* original);
} // namespace webrtc