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https://github.com/oxen-io/session-android.git
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83 lines
2.5 KiB
C
83 lines
2.5 KiB
C
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/*
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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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
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#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
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#include "webrtc/modules/audio_coding/main/test/Channel.h"
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#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class Config;
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class TestPack : public AudioPacketizationCallback {
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public:
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TestPack();
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~TestPack();
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void RegisterReceiverACM(AudioCodingModule* acm);
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int32_t SendData(FrameType frame_type, uint8_t payload_type,
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uint32_t timestamp, const uint8_t* payload_data,
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uint16_t payload_size,
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const RTPFragmentationHeader* fragmentation);
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uint16_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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AudioCodingModule* receiver_acm_;
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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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uint16_t payload_size_;
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};
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class TestAllCodecs : public ACMTest {
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public:
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explicit TestAllCodecs(int test_mode);
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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 side, char* codec_name, int32_t sampling_freq_hz,
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int rate, int packet_size, int extra_byte);
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void Run(TestPack* channel);
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void OpenOutFile(int test_number);
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void DisplaySendReceiveCodec();
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int test_mode_;
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scoped_ptr<AudioCodingModule> acm_a_;
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scoped_ptr<AudioCodingModule> acm_b_;
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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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int packet_size_bytes_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
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