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64437e8cc0
Calculate the RMS audio level of audio packets being sent before invoking an encoded frame transform, and pass them with the encode frame object. Before this, the audio level was calculated at send time by having rms_levels_ look at all audio samples encoded since the last send. This is fine without a transform, as this is done synchronously after encoding, but with an async transform which might take arbitrarily long, we could end up marking older audio packets with newer audio levels, or not at all. This also makes things work correctly if external encoded frames are injected from elsewhere to be sent, and exposes the AudioLevel on the TransformableFrame interface. Bug: chromium:337193823, webrtc:42226202 Change-Id: If55d2c1d30dc03408ca9fb0193d791db44428316 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349263 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#42193}
92 lines
3.6 KiB
C++
92 lines
3.6 KiB
C++
/*
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* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_
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#define AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_
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#include <memory>
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#include <string>
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#include "api/frame_transformer_interface.h"
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#include "api/sequence_checker.h"
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#include "api/task_queue/task_queue_base.h"
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#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/synchronization/mutex.h"
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namespace webrtc {
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// Delegates calls to FrameTransformerInterface to transform frames, and to
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// ChannelSend to send the transformed frames using `send_frame_callback_` on
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// the `encoder_queue_`.
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// OnTransformedFrame() can be called from any thread, the delegate ensures
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// thread-safe access to the ChannelSend callback.
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class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback {
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public:
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using SendFrameCallback =
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std::function<int32_t(AudioFrameType frameType,
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uint8_t payloadType,
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uint32_t rtp_timestamp_with_offset,
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rtc::ArrayView<const uint8_t> payload,
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int64_t absolute_capture_timestamp_ms,
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rtc::ArrayView<const uint32_t> csrcs,
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absl::optional<uint8_t> audio_level_dbov)>;
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ChannelSendFrameTransformerDelegate(
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SendFrameCallback send_frame_callback,
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
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TaskQueueBase* encoder_queue);
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// Registers `this` as callback for `frame_transformer_`, to get the
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// transformed frames.
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void Init();
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// Unregisters and releases the `frame_transformer_` reference, and resets
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// `send_frame_callback_` under lock. Called from ChannelSend destructor to
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// prevent running the callback on a dangling channel.
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void Reset();
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// Delegates the call to FrameTransformerInterface::TransformFrame, to
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// transform the frame asynchronously.
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void Transform(AudioFrameType frame_type,
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uint8_t payload_type,
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uint32_t rtp_timestamp,
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const uint8_t* payload_data,
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size_t payload_size,
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int64_t absolute_capture_timestamp_ms,
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uint32_t ssrc,
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const std::string& codec_mime_type,
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absl::optional<uint8_t> audio_level_dbov);
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// Implements TransformedFrameCallback. Can be called on any thread.
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void OnTransformedFrame(
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std::unique_ptr<TransformableFrameInterface> frame) override;
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void StartShortCircuiting() override;
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// Delegates the call to ChannelSend::SendRtpAudio on the `encoder_queue_`,
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// by calling `send_audio_callback_`.
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void SendFrame(std::unique_ptr<TransformableFrameInterface> frame) const;
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protected:
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~ChannelSendFrameTransformerDelegate() override = default;
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private:
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mutable Mutex send_lock_;
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SendFrameCallback send_frame_callback_ RTC_GUARDED_BY(send_lock_);
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_;
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TaskQueueBase* const encoder_queue_;
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bool short_circuit_ RTC_GUARDED_BY(send_lock_) = false;
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};
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std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
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TransformableAudioFrameInterface* original);
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} // namespace webrtc
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#endif // AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_
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