webrtc_m130/test/mock_audio_encoder.h

107 lines
4.0 KiB
C
Raw Normal View History

/*
* Copyright (c) 2014 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 TEST_MOCK_AUDIO_ENCODER_H_
#define TEST_MOCK_AUDIO_ENCODER_H_
#include <string>
#include "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "test/gmock.h"
namespace webrtc {
class MockAudioEncoder : public AudioEncoder {
public:
// TODO(nisse): Valid overrides commented out, because the gmock
// methods don't use any override declarations, and we want to avoid
// warnings from -Winconsistent-missing-override. See
// http://crbug.com/428099.
MockAudioEncoder();
~MockAudioEncoder();
MOCK_METHOD1(Mark, void(std::string desc));
MOCK_CONST_METHOD0(SampleRateHz, int());
MOCK_CONST_METHOD0(NumChannels, size_t());
MOCK_CONST_METHOD0(RtpTimestampRateHz, int());
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
MOCK_CONST_METHOD0(GetTargetBitrate, int());
MOCK_METHOD0(Reset, void());
MOCK_METHOD1(SetFec, bool(bool enable));
MOCK_METHOD1(SetDtx, bool(bool enable));
MOCK_METHOD1(SetApplication, bool(Application application));
MOCK_METHOD1(SetMaxPlaybackRate, void(int frequency_hz));
MOCK_METHOD1(SetMaxBitrate, void(int max_bps));
MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
MOCK_METHOD2(OnReceivedUplinkBandwidth,
void(int target_audio_bitrate_bps,
absl::optional<int64_t> probing_interval_ms));
MOCK_METHOD1(OnReceivedUplinkPacketLossFraction,
void(float uplink_packet_loss_fraction));
MOCK_METHOD2(EnableAudioNetworkAdaptor,
bool(const std::string& config_string, RtcEventLog* event_log));
// Note, we explicitly chose not to create a mock for the Encode method.
MOCK_METHOD3(EncodeImpl,
EncodedInfo(uint32_t timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded));
class FakeEncoding {
public:
// Creates a functor that will return |info| and adjust the rtc::Buffer
// given as input to it, so it is info.encoded_bytes larger.
explicit FakeEncoding(const AudioEncoder::EncodedInfo& info);
// Shorthand version of the constructor above, for when only setting
// encoded_bytes in the EncodedInfo object matters.
explicit FakeEncoding(size_t encoded_bytes);
AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded);
private:
AudioEncoder::EncodedInfo info_;
};
class CopyEncoding {
public:
~CopyEncoding();
// Creates a functor that will return |info| and append the data in the
// payload to the buffer given as input to it. Up to info.encoded_bytes are
// appended - make sure the payload is big enough! Since it uses an
// ArrayView, it _does not_ copy the payload. Make sure it doesn't fall out
// of scope!
CopyEncoding(AudioEncoder::EncodedInfo info,
rtc::ArrayView<const uint8_t> payload);
// Shorthand version of the constructor above, for when you wish to append
// the whole payload and do not care about any EncodedInfo attribute other
// than encoded_bytes.
explicit CopyEncoding(rtc::ArrayView<const uint8_t> payload);
AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded);
private:
AudioEncoder::EncodedInfo info_;
rtc::ArrayView<const uint8_t> payload_;
};
};
} // namespace webrtc
#endif // TEST_MOCK_AUDIO_ENCODER_H_