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665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
64 lines
2.1 KiB
C++
64 lines
2.1 KiB
C++
/*
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* Copyright (c) 2015 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 MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
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#include <memory>
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#include <string>
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#include "api/audio_codecs/audio_decoder.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/audio_coding/neteq/include/neteq.h"
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namespace webrtc {
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namespace test {
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// This test class provides a way run NetEQ with an external decoder.
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class NetEqExternalDecoderTest {
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protected:
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static const uint8_t kPayloadType = 95;
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static const int kOutputLengthMs = 10;
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// The external decoder |decoder| is suppose to be of type |codec|.
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NetEqExternalDecoderTest(NetEqDecoder codec,
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int sample_rate_hz,
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AudioDecoder* decoder);
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virtual ~NetEqExternalDecoderTest() {}
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// In Init(), we register the external decoder.
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void Init();
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// Inserts a new packet with |rtp_header| and |payload| of
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// |payload_size_bytes| bytes. The |receive_timestamp| is an indication
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// of the time when the packet was received, and should be measured with
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// the same tick rate as the RTP timestamp of the current payload.
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virtual void InsertPacket(RTPHeader rtp_header,
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rtc::ArrayView<const uint8_t> payload,
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uint32_t receive_timestamp);
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// Get 10 ms of audio data.
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void GetOutputAudio(AudioFrame* output);
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NetEq* neteq() { return neteq_.get(); }
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private:
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NetEqDecoder codec_;
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std::string name_ = "dummy name";
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AudioDecoder* decoder_;
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int sample_rate_hz_;
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size_t channels_;
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std::unique_ptr<NetEq> neteq_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
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