Michael Froman
209699b3f9
Upstream commit: https://webrtc.googlesource.com/src/+/97b46e12582606a238d4f0c8524365cf5bdcb411 Run tools_webrtc/apply-clang-tidy on the repo This uncovered the fact that its descent into third_party was not yet prevented; added running gclient sync -f to the script so that changes outside our repo would be reverted. Bug: None Change-Id: I18f4e75c57174b765ed941a062ed00613874f324 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/441061 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46602}
90 lines
2.5 KiB
C++
90 lines
2.5 KiB
C++
/*
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* Copyright (c) 2012 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_TEST_TESTALLCODECS_H_
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#define MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
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#include <cstddef>
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#include <cstdint>
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#include <memory>
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#include "api/environment/environment.h"
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
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#include "modules/audio_coding/test/PCMFile.h"
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namespace webrtc {
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class NetEq;
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class TestPack : public AudioPacketizationCallback {
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public:
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TestPack();
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~TestPack() override;
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void RegisterReceiverNetEq(NetEq* neteq);
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int32_t SendData(AudioFrameType frame_type,
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uint8_t payload_type,
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uint32_t timestamp,
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const uint8_t* payload_data,
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size_t payload_size,
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int64_t absolute_capture_timestamp_ms) override;
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size_t payload_size();
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uint32_t timestamp_diff();
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void reset_payload_size();
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private:
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NetEq* neteq_;
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uint16_t sequence_number_;
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uint8_t payload_data_[60 * 32 * 2 * 2];
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uint32_t timestamp_diff_;
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uint32_t last_in_timestamp_;
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uint64_t total_bytes_;
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size_t payload_size_;
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};
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class TestAllCodecs {
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public:
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TestAllCodecs();
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~TestAllCodecs();
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void Perform();
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private:
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// The default value of '-1' indicates that the registration is based only on
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// codec name, and a sampling frequency matching is not required.
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// This is useful for codecs which support several sampling frequency.
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// Note! Only mono mode is tested in this test.
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void RegisterSendCodec(char* codec_name,
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int32_t sampling_freq_hz,
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int rate,
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int packet_size,
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size_t extra_byte);
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void Run(TestPack* channel);
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void OpenOutFile(int test_number);
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const Environment env_;
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std::unique_ptr<AudioCodingModule> acm_a_;
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std::unique_ptr<NetEq> neteq_;
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TestPack* channel_a_to_b_;
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PCMFile infile_a_;
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PCMFile outfile_b_;
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int test_count_;
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int packet_size_samples_;
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size_t packet_size_bytes_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
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