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 TransformableAudioFrame : public TransformableFrameInterface {
19 public:
TransformableAudioFrame(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 TransformableAudioFrame(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 ~TransformableAudioFrame() 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 { return payload_type_; }
GetAbsoluteCaptureTimestampMs() const48 int64_t GetAbsoluteCaptureTimestampMs() const {
49 return absolute_capture_timestamp_ms_;
50 }
51
52 private:
53 AudioFrameType frame_type_;
54 uint8_t payload_type_;
55 uint32_t rtp_timestamp_;
56 uint32_t rtp_start_timestamp_;
57 rtc::Buffer payload_;
58 int64_t absolute_capture_timestamp_ms_;
59 uint32_t ssrc_;
60 };
61 } // namespace
62
ChannelSendFrameTransformerDelegate(SendFrameCallback send_frame_callback,rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,rtc::TaskQueue * encoder_queue)63 ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
64 SendFrameCallback send_frame_callback,
65 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
66 rtc::TaskQueue* encoder_queue)
67 : send_frame_callback_(send_frame_callback),
68 frame_transformer_(std::move(frame_transformer)),
69 encoder_queue_(encoder_queue) {}
70
Init()71 void ChannelSendFrameTransformerDelegate::Init() {
72 frame_transformer_->RegisterTransformedFrameCallback(
73 rtc::scoped_refptr<TransformedFrameCallback>(this));
74 }
75
Reset()76 void ChannelSendFrameTransformerDelegate::Reset() {
77 frame_transformer_->UnregisterTransformedFrameCallback();
78 frame_transformer_ = nullptr;
79
80 MutexLock lock(&send_lock_);
81 send_frame_callback_ = SendFrameCallback();
82 }
83
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)84 void ChannelSendFrameTransformerDelegate::Transform(
85 AudioFrameType frame_type,
86 uint8_t payload_type,
87 uint32_t rtp_timestamp,
88 uint32_t rtp_start_timestamp,
89 const uint8_t* payload_data,
90 size_t payload_size,
91 int64_t absolute_capture_timestamp_ms,
92 uint32_t ssrc) {
93 frame_transformer_->Transform(std::make_unique<TransformableAudioFrame>(
94 frame_type, payload_type, rtp_timestamp, rtp_start_timestamp,
95 payload_data, payload_size, absolute_capture_timestamp_ms, ssrc));
96 }
97
OnTransformedFrame(std::unique_ptr<TransformableFrameInterface> frame)98 void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
99 std::unique_ptr<TransformableFrameInterface> frame) {
100 MutexLock lock(&send_lock_);
101 if (!send_frame_callback_)
102 return;
103 rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate = this;
104 encoder_queue_->PostTask(
105 [delegate = std::move(delegate), frame = std::move(frame)]() mutable {
106 delegate->SendFrame(std::move(frame));
107 });
108 }
109
SendFrame(std::unique_ptr<TransformableFrameInterface> frame) const110 void ChannelSendFrameTransformerDelegate::SendFrame(
111 std::unique_ptr<TransformableFrameInterface> frame) const {
112 MutexLock lock(&send_lock_);
113 RTC_DCHECK_RUN_ON(encoder_queue_);
114 if (!send_frame_callback_)
115 return;
116 auto* transformed_frame = static_cast<TransformableAudioFrame*>(frame.get());
117 send_frame_callback_(transformed_frame->GetFrameType(),
118 transformed_frame->GetPayloadType(),
119 transformed_frame->GetTimestamp() -
120 transformed_frame->GetStartTimestamp(),
121 transformed_frame->GetData(),
122 transformed_frame->GetAbsoluteCaptureTimestampMs());
123 }
124
125 } // namespace webrtc
126