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/*
* Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class Config;
class TestPack : public AudioPacketizationCallback {
public:
TestPack();
~TestPack();
void RegisterReceiverACM(AudioCodingModule* acm);
int32_t SendData(FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) override;
size_t payload_size();
uint32_t timestamp_diff();
void reset_payload_size();
private:
AudioCodingModule* receiver_acm_;
uint16_t sequence_number_;
uint8_t payload_data_[60 * 32 * 2 * 2];
uint32_t timestamp_diff_;
uint32_t last_in_timestamp_;
uint64_t total_bytes_;
size_t payload_size_;
};
class TestAllCodecs : public ACMTest {
public:
explicit TestAllCodecs(int test_mode);
~TestAllCodecs();
void Perform() override;
private:
// The default value of '-1' indicates that the registration is based only on
// codec name, and a sampling frequency matching is not required.
// This is useful for codecs which support several sampling frequency.
// Note! Only mono mode is tested in this test.
void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz,
int rate, int packet_size, size_t extra_byte);
void Run(TestPack* channel);
void OpenOutFile(int test_number);
void DisplaySendReceiveCodec();
int test_mode_;
rtc::scoped_ptr<AudioCodingModule> acm_a_;
rtc::scoped_ptr<AudioCodingModule> acm_b_;
TestPack* channel_a_to_b_;
PCMFile infile_a_;
PCMFile outfile_b_;
int test_count_;
int packet_size_samples_;
size_t packet_size_bytes_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
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