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/*
* Copyright 2023 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 PC_TEST_MOCK_VOICE_MEDIA_RECEIVE_CHANNEL_INTERFACE_H_
#define PC_TEST_MOCK_VOICE_MEDIA_RECEIVE_CHANNEL_INTERFACE_H_
#include <cstdint>
#include <memory>
#include <optional>
#include <set>
#include <type_traits>
#include <vector>
#include "api/call/audio_sink.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_types.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "media/base/media_channel.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "test/gmock.h"
namespace webrtc {
class MockVoiceMediaReceiveChannelInterface
: public VoiceMediaReceiveChannelInterface {
public:
MockVoiceMediaReceiveChannelInterface() {
ON_CALL(*this, AsVoiceReceiveChannel).WillByDefault(testing::Return(this));
}
// VoiceMediaReceiveChannelInterface
MOCK_METHOD(bool,
SetReceiverParameters,
(const webrtc::AudioReceiverParameters& params),
(override));
MOCK_METHOD(RtpParameters,
GetRtpReceiverParameters,
(uint32_t ssrc),
(const, override));
MOCK_METHOD(std::vector<RtpSource>,
GetSources,
(uint32_t ssrc),
(const, override));
MOCK_METHOD(RtpParameters,
GetDefaultRtpReceiveParameters,
(),
(const, override));
MOCK_METHOD(void, SetPlayout, (bool playout), (override));
MOCK_METHOD(bool,
SetOutputVolume,
(uint32_t ssrc, double volume),
(override));
MOCK_METHOD(bool, SetDefaultOutputVolume, (double volume), (override));
MOCK_METHOD(void,
SetRawAudioSink,
(uint32_t ssrc, std::unique_ptr<AudioSinkInterface> sink),
(override));
MOCK_METHOD(void,
SetDefaultRawAudioSink,
(std::unique_ptr<AudioSinkInterface> sink),
(override));
MOCK_METHOD(bool,
GetStats,
(webrtc::VoiceMediaReceiveInfo * stats, bool reset_legacy),
(override));
MOCK_METHOD(::webrtc::RtcpMode, RtcpMode, (), (const, override));
MOCK_METHOD(void, SetRtcpMode, (::webrtc::RtcpMode mode), (override));
MOCK_METHOD(void, SetReceiveNackEnabled, (bool enabled), (override));
MOCK_METHOD(void, SetReceiveNonSenderRttEnabled, (bool enabled), (override));
// MediaReceiveChannelInterface
MOCK_METHOD(VideoMediaReceiveChannelInterface*,
AsVideoReceiveChannel,
(),
(override));
MOCK_METHOD(VoiceMediaReceiveChannelInterface*,
AsVoiceReceiveChannel,
(),
(override));
MOCK_METHOD(MediaType, media_type, (), (const, override));
MOCK_METHOD(bool,
AddRecvStream,
(const webrtc::StreamParams& sp),
(override));
MOCK_METHOD(bool, RemoveRecvStream, (uint32_t ssrc), (override));
MOCK_METHOD(void, ResetUnsignaledRecvStream, (), (override));
MOCK_METHOD(void,
SetInterface,
(webrtc::MediaChannelNetworkInterface * iface),
(override));
MOCK_METHOD(void,
OnPacketReceived,
(const RtpPacketReceived& packet),
(override));
MOCK_METHOD(std::optional<uint32_t>,
GetUnsignaledSsrc,
(),
(const, override));
MOCK_METHOD(void,
ChooseReceiverReportSsrc,
(const std::set<uint32_t>& choices),
(override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdatePending, (), (override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdateComplete, (), (override));
MOCK_METHOD(void,
SetFrameDecryptor,
(uint32_t ssrc,
scoped_refptr<FrameDecryptorInterface> frame_decryptor),
(override));
MOCK_METHOD(void,
SetDepacketizerToDecoderFrameTransformer,
(uint32_t ssrc,
scoped_refptr<FrameTransformerInterface> frame_transformer),
(override));
MOCK_METHOD(bool,
SetBaseMinimumPlayoutDelayMs,
(uint32_t ssrc, int delay_ms),
(override));
MOCK_METHOD(std::optional<int>,
GetBaseMinimumPlayoutDelayMs,
(uint32_t ssrc),
(const, override));
};
static_assert(!std::is_abstract_v<MockVoiceMediaReceiveChannelInterface>, "");
} // namespace webrtc
// Re-export symbols from the webrtc namespace for backwards compatibility.
// TODO(bugs.webrtc.org/4222596): Remove once all references are updated.
#ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES
namespace cricket {
using ::webrtc::MockVoiceMediaReceiveChannelInterface;
} // namespace cricket
#endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES
#endif // PC_TEST_MOCK_VOICE_MEDIA_RECEIVE_CHANNEL_INTERFACE_H_
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