webrtc_m130/audio/mock_voe_channel_proxy.h
Philipp Hancke bad99ab253 RTCP: implement reduced size RTCP for audio
reduced-size RTCP, i.e. not prefixing RTCP packets with either a sender report or receiver report has been implemented for a long time but only for video.

This CL adds it for audio as well. This reduces the size of audio NACKs (16 bytes, typically one NACK per packet) sent by not prefixing it with a receiver report (32 bytes).
Other packets are not affected as e.g. transport-cc feedback does not add a RR even though that is technically required.

The effect on NACK can be tested by running Chromium with
  --disable-webrtc-encryption --force-fieldtrials=WebRTC-FakeNetworkReceiveConfig/loss_percent:5/
against this fiddle negotiating audio nack:
https://jsfiddle.net/fippo/8ubtLnfx/1/

BUG=webrtc:340041654

Change-Id: I06fb94742ff1b6f9a464c404bfc53913f23498d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350269
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42330}
2024-05-16 18:24:10 +00:00

190 lines
7.1 KiB
C++

/*
* Copyright (c) 2015 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.
*/
#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/crypto/frame_decryptor_interface.h"
#include "api/test/mock_frame_encryptor.h"
#include "audio/channel_receive.h"
#include "audio/channel_send.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "test/gmock.h"
namespace webrtc {
namespace test {
class MockChannelReceive : public voe::ChannelReceiveInterface {
public:
MOCK_METHOD(void, SetNACKStatus, (bool enable, int max_packets), (override));
MOCK_METHOD(void, SetRtcpMode, (RtcpMode mode), (override));
MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override));
MOCK_METHOD(void,
RegisterReceiverCongestionControlObjects,
(PacketRouter*),
(override));
MOCK_METHOD(void, ResetReceiverCongestionControlObjects, (), (override));
MOCK_METHOD(CallReceiveStatistics, GetRTCPStatistics, (), (const, override));
MOCK_METHOD(NetworkStatistics,
GetNetworkStatistics,
(bool),
(const, override));
MOCK_METHOD(AudioDecodingCallStats,
GetDecodingCallStatistics,
(),
(const, override));
MOCK_METHOD(int, GetSpeechOutputLevelFullRange, (), (const, override));
MOCK_METHOD(double, GetTotalOutputEnergy, (), (const, override));
MOCK_METHOD(double, GetTotalOutputDuration, (), (const, override));
MOCK_METHOD(uint32_t, GetDelayEstimate, (), (const, override));
MOCK_METHOD(void, SetSink, (AudioSinkInterface*), (override));
MOCK_METHOD(void, OnRtpPacket, (const RtpPacketReceived& packet), (override));
MOCK_METHOD(void,
ReceivedRTCPPacket,
(const uint8_t*, size_t length),
(override));
MOCK_METHOD(void, SetChannelOutputVolumeScaling, (float scaling), (override));
MOCK_METHOD(AudioMixer::Source::AudioFrameInfo,
GetAudioFrameWithInfo,
(int sample_rate_hz, AudioFrame*),
(override));
MOCK_METHOD(int, PreferredSampleRate, (), (const, override));
MOCK_METHOD(void, SetSourceTracker, (SourceTracker*), (override));
MOCK_METHOD(void,
SetAssociatedSendChannel,
(const voe::ChannelSendInterface*),
(override));
MOCK_METHOD(bool,
GetPlayoutRtpTimestamp,
(uint32_t*, int64_t*),
(const, override));
MOCK_METHOD(void,
SetEstimatedPlayoutNtpTimestampMs,
(int64_t ntp_timestamp_ms, int64_t time_ms),
(override));
MOCK_METHOD(absl::optional<int64_t>,
GetCurrentEstimatedPlayoutNtpTimestampMs,
(int64_t now_ms),
(const, override));
MOCK_METHOD(absl::optional<Syncable::Info>,
GetSyncInfo,
(),
(const, override));
MOCK_METHOD(bool, SetMinimumPlayoutDelay, (int delay_ms), (override));
MOCK_METHOD(bool, SetBaseMinimumPlayoutDelayMs, (int delay_ms), (override));
MOCK_METHOD(int, GetBaseMinimumPlayoutDelayMs, (), (const, override));
MOCK_METHOD((absl::optional<std::pair<int, SdpAudioFormat>>),
GetReceiveCodec,
(),
(const, override));
MOCK_METHOD(void,
SetReceiveCodecs,
((const std::map<int, SdpAudioFormat>& codecs)),
(override));
MOCK_METHOD(void, StartPlayout, (), (override));
MOCK_METHOD(void, StopPlayout, (), (override));
MOCK_METHOD(
void,
SetDepacketizerToDecoderFrameTransformer,
(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
(override));
MOCK_METHOD(
void,
SetFrameDecryptor,
(rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
(override));
MOCK_METHOD(void, OnLocalSsrcChange, (uint32_t local_ssrc), (override));
MOCK_METHOD(uint32_t, GetLocalSsrc, (), (const, override));
};
class MockChannelSend : public voe::ChannelSendInterface {
public:
MOCK_METHOD(void,
SetEncoder,
(int payload_type,
const SdpAudioFormat& encoder_format,
std::unique_ptr<AudioEncoder> encoder),
(override));
MOCK_METHOD(
void,
ModifyEncoder,
(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier),
(override));
MOCK_METHOD(void,
CallEncoder,
(rtc::FunctionView<void(AudioEncoder*)> modifier),
(override));
MOCK_METHOD(void, SetRTCP_CNAME, (absl::string_view c_name), (override));
MOCK_METHOD(void,
SetSendAudioLevelIndicationStatus,
(bool enable, int id),
(override));
MOCK_METHOD(void,
RegisterSenderCongestionControlObjects,
(RtpTransportControllerSendInterface*),
(override));
MOCK_METHOD(void, ResetSenderCongestionControlObjects, (), (override));
MOCK_METHOD(CallSendStatistics, GetRTCPStatistics, (), (const, override));
MOCK_METHOD(std::vector<ReportBlockData>,
GetRemoteRTCPReportBlocks,
(),
(const, override));
MOCK_METHOD(ANAStats, GetANAStatistics, (), (const, override));
MOCK_METHOD(void,
RegisterCngPayloadType,
(int payload_type, int payload_frequency),
(override));
MOCK_METHOD(void,
SetSendTelephoneEventPayloadType,
(int payload_type, int payload_frequency),
(override));
MOCK_METHOD(bool,
SendTelephoneEventOutband,
(int event, int duration_ms),
(override));
MOCK_METHOD(void,
OnBitrateAllocation,
(BitrateAllocationUpdate update),
(override));
MOCK_METHOD(void, SetInputMute, (bool muted), (override));
MOCK_METHOD(void,
ReceivedRTCPPacket,
(const uint8_t*, size_t length),
(override));
MOCK_METHOD(void,
ProcessAndEncodeAudio,
(std::unique_ptr<AudioFrame>),
(override));
MOCK_METHOD(RtpRtcpInterface*, GetRtpRtcp, (), (const, override));
MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
MOCK_METHOD(int64_t, GetRTT, (), (const, override));
MOCK_METHOD(void, StartSend, (), (override));
MOCK_METHOD(void, StopSend, (), (override));
MOCK_METHOD(void,
SetFrameEncryptor,
(rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor),
(override));
MOCK_METHOD(
void,
SetEncoderToPacketizerFrameTransformer,
(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
(override));
};
} // namespace test
} // namespace webrtc
#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_