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62aee9379c
This adds the ability to disable legacy overhead calculation so we'll use the available data on per packet over head and frame length range to set the min and max total allocatable bitrate. Bug: webrtc:11001 Change-Id: I2a94499433e15bad11a08f81fe7f1dfc27982cdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155175 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29368}
109 lines
4.1 KiB
C++
109 lines
4.1 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef TEST_MOCK_AUDIO_ENCODER_H_
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#define TEST_MOCK_AUDIO_ENCODER_H_
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#include <string>
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#include "api/array_view.h"
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#include "api/audio_codecs/audio_encoder.h"
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#include "test/gmock.h"
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namespace webrtc {
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class MockAudioEncoder : public AudioEncoder {
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public:
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// TODO(nisse): Valid overrides commented out, because the gmock
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// methods don't use any override declarations, and we want to avoid
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// warnings from -Winconsistent-missing-override. See
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// http://crbug.com/428099.
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MockAudioEncoder();
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~MockAudioEncoder();
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MOCK_METHOD1(Mark, void(std::string desc));
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MOCK_CONST_METHOD0(SampleRateHz, int());
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MOCK_CONST_METHOD0(NumChannels, size_t());
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MOCK_CONST_METHOD0(RtpTimestampRateHz, int());
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MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
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MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
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MOCK_CONST_METHOD0(GetTargetBitrate, int());
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MOCK_CONST_METHOD0(GetFrameLengthRange,
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absl::optional<std::pair<TimeDelta, TimeDelta>>());
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MOCK_METHOD0(Reset, void());
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MOCK_METHOD1(SetFec, bool(bool enable));
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MOCK_METHOD1(SetDtx, bool(bool enable));
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MOCK_METHOD1(SetApplication, bool(Application application));
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MOCK_METHOD1(SetMaxPlaybackRate, void(int frequency_hz));
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MOCK_METHOD1(SetMaxBitrate, void(int max_bps));
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MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
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MOCK_METHOD2(OnReceivedUplinkBandwidth,
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void(int target_audio_bitrate_bps,
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absl::optional<int64_t> probing_interval_ms));
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MOCK_METHOD1(OnReceivedUplinkPacketLossFraction,
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void(float uplink_packet_loss_fraction));
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MOCK_METHOD2(EnableAudioNetworkAdaptor,
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bool(const std::string& config_string, RtcEventLog* event_log));
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// Note, we explicitly chose not to create a mock for the Encode method.
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MOCK_METHOD3(EncodeImpl,
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EncodedInfo(uint32_t timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded));
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class FakeEncoding {
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public:
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// Creates a functor that will return |info| and adjust the rtc::Buffer
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// given as input to it, so it is info.encoded_bytes larger.
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explicit FakeEncoding(const AudioEncoder::EncodedInfo& info);
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// Shorthand version of the constructor above, for when only setting
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// encoded_bytes in the EncodedInfo object matters.
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explicit FakeEncoding(size_t encoded_bytes);
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AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded);
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private:
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AudioEncoder::EncodedInfo info_;
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};
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class CopyEncoding {
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public:
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~CopyEncoding();
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// Creates a functor that will return |info| and append the data in the
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// payload to the buffer given as input to it. Up to info.encoded_bytes are
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// appended - make sure the payload is big enough! Since it uses an
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// ArrayView, it _does not_ copy the payload. Make sure it doesn't fall out
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// of scope!
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CopyEncoding(AudioEncoder::EncodedInfo info,
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rtc::ArrayView<const uint8_t> payload);
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// Shorthand version of the constructor above, for when you wish to append
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// the whole payload and do not care about any EncodedInfo attribute other
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// than encoded_bytes.
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explicit CopyEncoding(rtc::ArrayView<const uint8_t> payload);
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AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded);
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private:
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AudioEncoder::EncodedInfo info_;
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rtc::ArrayView<const uint8_t> payload_;
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};
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};
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} // namespace webrtc
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#endif // TEST_MOCK_AUDIO_ENCODER_H_
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