webrtc/audio/channel_send_frame_transformer_delegate_unittest.cc
Tony Herre 64437e8cc0 Calculate the audio level of audio packets before encoded transforms
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}
2024-04-29 15:14:25 +00:00

274 lines
12 KiB
C++

/*
* 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 "audio/channel_send_frame_transformer_delegate.h"
#include <memory>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/test/mock_frame_transformer.h"
#include "api/test/mock_transformable_audio_frame.h"
#include "rtc_base/task_queue_for_test.h"
#include "test/gmock.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
using ::testing::_;
using ::testing::ElementsAre;
using ::testing::ElementsAreArray;
using ::testing::NiceMock;
using ::testing::Optional;
using ::testing::Return;
using ::testing::SaveArg;
const uint8_t mock_data[] = {1, 2, 3, 4};
class MockChannelSend {
public:
MockChannelSend() = default;
~MockChannelSend() = default;
MOCK_METHOD(int32_t,
SendFrame,
(AudioFrameType frameType,
uint8_t payloadType,
uint32_t rtp_timestamp,
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level_dbov));
ChannelSendFrameTransformerDelegate::SendFrameCallback callback() {
return [this](AudioFrameType frameType, uint8_t payloadType,
uint32_t rtp_timestamp, rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms,
rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level_dbov) {
return SendFrame(frameType, payloadType, rtp_timestamp, payload,
absolute_capture_timestamp_ms, csrcs, audio_level_dbov);
};
}
};
std::unique_ptr<TransformableAudioFrameInterface> CreateMockReceiverFrame(
const std::vector<uint32_t>& csrcs,
absl::optional<uint8_t> audio_level_dbov) {
std::unique_ptr<MockTransformableAudioFrame> mock_frame =
std::make_unique<NiceMock<MockTransformableAudioFrame>>();
rtc::ArrayView<const uint8_t> payload(mock_data);
ON_CALL(*mock_frame, GetData).WillByDefault(Return(payload));
ON_CALL(*mock_frame, GetPayloadType).WillByDefault(Return(0));
ON_CALL(*mock_frame, GetDirection)
.WillByDefault(Return(TransformableFrameInterface::Direction::kReceiver));
ON_CALL(*mock_frame, GetContributingSources).WillByDefault(Return(csrcs));
ON_CALL(*mock_frame, SequenceNumber).WillByDefault(Return(987654321));
ON_CALL(*mock_frame, AudioLevel).WillByDefault(Return(audio_level_dbov));
return mock_frame;
}
std::unique_ptr<TransformableAudioFrameInterface> CreateFrame() {
TaskQueueForTest channel_queue("channel_queue");
rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
MockChannelSend mock_channel;
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
std::unique_ptr<TransformableFrameInterface> frame;
ON_CALL(*mock_frame_transformer, Transform)
.WillByDefault(
[&frame](
std::unique_ptr<TransformableFrameInterface> transform_frame) {
frame = std::move(transform_frame);
});
delegate->Transform(
AudioFrameType::kEmptyFrame, 0, 0, mock_data, sizeof(mock_data), 0,
/*ssrc=*/0, /*mimeType=*/"audio/opus", /*audio_level_dbov=*/123);
return absl::WrapUnique(
static_cast<webrtc::TransformableAudioFrameInterface*>(frame.release()));
}
// Test that the delegate registers itself with the frame transformer on Init().
TEST(ChannelSendFrameTransformerDelegateTest,
RegisterTransformedFrameCallbackOnInit) {
rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
rtc::make_ref_counted<MockFrameTransformer>();
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
ChannelSendFrameTransformerDelegate::SendFrameCallback(),
mock_frame_transformer, nullptr);
EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback);
delegate->Init();
}
// Test that the delegate unregisters itself from the frame transformer on
// Reset().
TEST(ChannelSendFrameTransformerDelegateTest,
UnregisterTransformedFrameCallbackOnReset) {
rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
rtc::make_ref_counted<MockFrameTransformer>();
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
ChannelSendFrameTransformerDelegate::SendFrameCallback(),
mock_frame_transformer, nullptr);
EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
delegate->Reset();
}
// Test that when the delegate receives a transformed frame from the frame
// transformer, it passes it to the channel using the SendFrameCallback.
TEST(ChannelSendFrameTransformerDelegateTest,
TransformRunsChannelSendCallback) {
TaskQueueForTest channel_queue("channel_queue");
rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
MockChannelSend mock_channel;
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
rtc::scoped_refptr<TransformedFrameCallback> callback;
EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
.WillOnce(SaveArg<0>(&callback));
delegate->Init();
ASSERT_TRUE(callback);
const uint8_t data[] = {1, 2, 3, 4};
EXPECT_CALL(mock_channel, SendFrame);
ON_CALL(*mock_frame_transformer, Transform)
.WillByDefault(
[&callback](std::unique_ptr<TransformableFrameInterface> frame) {
callback->OnTransformedFrame(std::move(frame));
});
delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, data, sizeof(data), 0,
/*ssrc=*/0, /*mimeType=*/"audio/opus",
/*audio_level_dbov=*/31);
channel_queue.WaitForPreviouslyPostedTasks();
}
// Test that when the delegate receives a Incoming frame from the frame
// transformer, it passes it to the channel using the SendFrameCallback.
TEST(ChannelSendFrameTransformerDelegateTest,
TransformRunsChannelSendCallbackForIncomingFrame) {
TaskQueueForTest channel_queue("channel_queue");
rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
MockChannelSend mock_channel;
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
rtc::scoped_refptr<TransformedFrameCallback> callback;
EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
.WillOnce(SaveArg<0>(&callback));
delegate->Init();
ASSERT_TRUE(callback);
const std::vector<uint32_t> csrcs = {123, 234, 345, 456};
const uint8_t audio_level_dbov = 17;
EXPECT_CALL(mock_channel, SendFrame).Times(0);
EXPECT_CALL(mock_channel,
SendFrame(_, 0, 0, ElementsAreArray(mock_data), _,
ElementsAreArray(csrcs), Optional(audio_level_dbov)));
ON_CALL(*mock_frame_transformer, Transform)
.WillByDefault([&](std::unique_ptr<TransformableFrameInterface> frame) {
callback->OnTransformedFrame(CreateMockReceiverFrame(
csrcs, absl::optional<uint8_t>(audio_level_dbov)));
});
delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, mock_data,
sizeof(mock_data), 0,
/*ssrc=*/0, /*mimeType=*/"audio/opus",
/*audio_level_dbov=*/absl::nullopt);
channel_queue.WaitForPreviouslyPostedTasks();
}
// Test that if the delegate receives a transformed frame after it has been
// reset, it does not run the SendFrameCallback, as the channel is destroyed
// after resetting the delegate.
TEST(ChannelSendFrameTransformerDelegateTest,
OnTransformedDoesNotRunChannelSendCallbackAfterReset) {
TaskQueueForTest channel_queue("channel_queue");
rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
rtc::make_ref_counted<testing::NiceMock<MockFrameTransformer>>();
MockChannelSend mock_channel;
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
delegate->Reset();
EXPECT_CALL(mock_channel, SendFrame).Times(0);
delegate->OnTransformedFrame(std::make_unique<MockTransformableAudioFrame>());
channel_queue.WaitForPreviouslyPostedTasks();
}
TEST(ChannelSendFrameTransformerDelegateTest, ShortCircuitingSkipsTransform) {
TaskQueueForTest channel_queue("channel_queue");
rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
rtc::make_ref_counted<testing::NiceMock<MockFrameTransformer>>();
MockChannelSend mock_channel;
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
mock_channel.callback(), mock_frame_transformer, channel_queue.Get());
delegate->StartShortCircuiting();
// Will not call the actual transformer.
EXPECT_CALL(*mock_frame_transformer, Transform).Times(0);
// Will pass the frame straight to the channel.
EXPECT_CALL(mock_channel, SendFrame);
const uint8_t data[] = {1, 2, 3, 4};
delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, data, sizeof(data), 0,
/*ssrc=*/0, /*mimeType=*/"audio/opus",
/*audio_level_dbov=*/absl::nullopt);
}
TEST(ChannelSendFrameTransformerDelegateTest,
CloningSenderFramePreservesInformation) {
std::unique_ptr<TransformableAudioFrameInterface> frame = CreateFrame();
std::unique_ptr<TransformableAudioFrameInterface> cloned_frame =
CloneSenderAudioFrame(frame.get());
EXPECT_EQ(cloned_frame->GetTimestamp(), frame->GetTimestamp());
EXPECT_EQ(cloned_frame->GetSsrc(), frame->GetSsrc());
EXPECT_EQ(cloned_frame->Type(), frame->Type());
EXPECT_EQ(cloned_frame->GetPayloadType(), frame->GetPayloadType());
EXPECT_EQ(cloned_frame->GetMimeType(), frame->GetMimeType());
EXPECT_THAT(cloned_frame->GetContributingSources(),
ElementsAreArray(frame->GetContributingSources()));
EXPECT_EQ(cloned_frame->AudioLevel(), frame->AudioLevel());
}
TEST(ChannelSendFrameTransformerDelegateTest, CloningReceiverFrameWithCsrcs) {
std::unique_ptr<TransformableAudioFrameInterface> frame =
CreateMockReceiverFrame(/*csrcs=*/{123, 234, 345},
absl::optional<uint8_t>(72));
std::unique_ptr<TransformableAudioFrameInterface> cloned_frame =
CloneSenderAudioFrame(frame.get());
EXPECT_EQ(cloned_frame->GetTimestamp(), frame->GetTimestamp());
EXPECT_EQ(cloned_frame->GetSsrc(), frame->GetSsrc());
EXPECT_EQ(cloned_frame->Type(), frame->Type());
EXPECT_EQ(cloned_frame->GetPayloadType(), frame->GetPayloadType());
EXPECT_EQ(cloned_frame->GetMimeType(), frame->GetMimeType());
EXPECT_EQ(cloned_frame->AbsoluteCaptureTimestamp(),
frame->AbsoluteCaptureTimestamp());
ASSERT_NE(frame->GetContributingSources().size(), 0u);
EXPECT_THAT(cloned_frame->GetContributingSources(),
ElementsAreArray(frame->GetContributingSources()));
EXPECT_EQ(cloned_frame->SequenceNumber(), frame->SequenceNumber());
EXPECT_EQ(cloned_frame->AudioLevel(), frame->AudioLevel());
}
} // namespace
} // namespace webrtc