1 /*
2 * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "audio/channel_send_frame_transformer_delegate.h"
12
13 #include <utility>
14
15 namespace webrtc {
16 namespace {
17
18 class TransformableOutgoingAudioFrame : public TransformableFrameInterface {
19 public:
TransformableOutgoingAudioFrame(AudioFrameType frame_type,uint8_t payload_type,uint32_t rtp_timestamp,uint32_t rtp_start_timestamp,const uint8_t * payload_data,size_t payload_size,int64_t absolute_capture_timestamp_ms,uint32_t ssrc)20 TransformableOutgoingAudioFrame(AudioFrameType frame_type,
21 uint8_t payload_type,
22 uint32_t rtp_timestamp,
23 uint32_t rtp_start_timestamp,
24 const uint8_t* payload_data,
25 size_t payload_size,
26 int64_t absolute_capture_timestamp_ms,
27 uint32_t ssrc)
28 : frame_type_(frame_type),
29 payload_type_(payload_type),
30 rtp_timestamp_(rtp_timestamp),
31 rtp_start_timestamp_(rtp_start_timestamp),
32 payload_(payload_data, payload_size),
33 absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms),
34 ssrc_(ssrc) {}
35 ~TransformableOutgoingAudioFrame() override = default;
GetData() const36 rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
SetData(rtc::ArrayView<const uint8_t> data)37 void SetData(rtc::ArrayView<const uint8_t> data) override {
38 payload_.SetData(data.data(), data.size());
39 }
GetTimestamp() const40 uint32_t GetTimestamp() const override {
41 return rtp_timestamp_ + rtp_start_timestamp_;
42 }
GetStartTimestamp() const43 uint32_t GetStartTimestamp() const { return rtp_start_timestamp_; }
GetSsrc() const44 uint32_t GetSsrc() const override { return ssrc_; }
45
GetFrameType() const46 AudioFrameType GetFrameType() const { return frame_type_; }
GetPayloadType() const47 uint8_t GetPayloadType() const override { return payload_type_; }
GetAbsoluteCaptureTimestampMs() const48 int64_t GetAbsoluteCaptureTimestampMs() const {
49 return absolute_capture_timestamp_ms_;
50 }
GetDirection() const51 Direction GetDirection() const override { return Direction::kSender; }
52
53 private:
54 AudioFrameType frame_type_;
55 uint8_t payload_type_;
56 uint32_t rtp_timestamp_;
57 uint32_t rtp_start_timestamp_;
58 rtc::Buffer payload_;
59 int64_t absolute_capture_timestamp_ms_;
60 uint32_t ssrc_;
61 };
62 } // namespace
63
ChannelSendFrameTransformerDelegate(SendFrameCallback send_frame_callback,rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,rtc::TaskQueue * encoder_queue)64 ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
65 SendFrameCallback send_frame_callback,
66 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
67 rtc::TaskQueue* encoder_queue)
68 : send_frame_callback_(send_frame_callback),
69 frame_transformer_(std::move(frame_transformer)),
70 encoder_queue_(encoder_queue) {}
71
Init()72 void ChannelSendFrameTransformerDelegate::Init() {
73 frame_transformer_->RegisterTransformedFrameCallback(
74 rtc::scoped_refptr<TransformedFrameCallback>(this));
75 }
76
Reset()77 void ChannelSendFrameTransformerDelegate::Reset() {
78 frame_transformer_->UnregisterTransformedFrameCallback();
79 frame_transformer_ = nullptr;
80
81 MutexLock lock(&send_lock_);
82 send_frame_callback_ = SendFrameCallback();
83 }
84
Transform(AudioFrameType frame_type,uint8_t payload_type,uint32_t rtp_timestamp,uint32_t rtp_start_timestamp,const uint8_t * payload_data,size_t payload_size,int64_t absolute_capture_timestamp_ms,uint32_t ssrc)85 void ChannelSendFrameTransformerDelegate::Transform(
86 AudioFrameType frame_type,
87 uint8_t payload_type,
88 uint32_t rtp_timestamp,
89 uint32_t rtp_start_timestamp,
90 const uint8_t* payload_data,
91 size_t payload_size,
92 int64_t absolute_capture_timestamp_ms,
93 uint32_t ssrc) {
94 frame_transformer_->Transform(
95 std::make_unique<TransformableOutgoingAudioFrame>(
96 frame_type, payload_type, rtp_timestamp, rtp_start_timestamp,
97 payload_data, payload_size, absolute_capture_timestamp_ms, ssrc));
98 }
99
OnTransformedFrame(std::unique_ptr<TransformableFrameInterface> frame)100 void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
101 std::unique_ptr<TransformableFrameInterface> frame) {
102 MutexLock lock(&send_lock_);
103 if (!send_frame_callback_)
104 return;
105 rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate(this);
106 encoder_queue_->PostTask(
107 [delegate = std::move(delegate), frame = std::move(frame)]() mutable {
108 delegate->SendFrame(std::move(frame));
109 });
110 }
111
SendFrame(std::unique_ptr<TransformableFrameInterface> frame) const112 void ChannelSendFrameTransformerDelegate::SendFrame(
113 std::unique_ptr<TransformableFrameInterface> frame) const {
114 MutexLock lock(&send_lock_);
115 RTC_DCHECK_RUN_ON(encoder_queue_);
116 RTC_CHECK_EQ(frame->GetDirection(),
117 TransformableFrameInterface::Direction::kSender);
118 if (!send_frame_callback_)
119 return;
120 auto* transformed_frame =
121 static_cast<TransformableOutgoingAudioFrame*>(frame.get());
122 send_frame_callback_(transformed_frame->GetFrameType(),
123 transformed_frame->GetPayloadType(),
124 transformed_frame->GetTimestamp() -
125 transformed_frame->GetStartTimestamp(),
126 transformed_frame->GetData(),
127 transformed_frame->GetAbsoluteCaptureTimestampMs());
128 }
129
130 } // namespace webrtc
131