chromium/third_party/webrtc/modules/audio_coding/test/TestStereo.cc

/*
 *  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.
 */

#include "modules/audio_coding/test/TestStereo.h"

#include <string>

#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment_factory.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"

namespace webrtc {

// Class for simulating packet handling
TestPackStereo::TestPackStereo()
    :{}

TestPackStereo::~TestPackStereo() {}

void TestPackStereo::RegisterReceiverACM(acm2::AcmReceiver* acm_receiver) {}

int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
                                 const uint8_t payload_type,
                                 const uint32_t timestamp,
                                 const uint8_t* payload_data,
                                 const size_t payload_size,
                                 int64_t absolute_capture_timestamp_ms) {}

uint16_t TestPackStereo::payload_size() {}

uint32_t TestPackStereo::timestamp_diff() {}

void TestPackStereo::reset_payload_size() {}

void TestPackStereo::set_codec_mode(enum StereoMonoMode mode) {}

void TestPackStereo::set_lost_packet(bool lost) {}

TestStereo::TestStereo()
    :{}

TestStereo::~TestStereo() {}

void TestStereo::Perform() {}

// Register Codec to use in the test
//
// Input:   side             - which ACM to use, 'A' or 'B'
//          codec_name       - name to use when register the codec
//          sampling_freq_hz - sampling frequency in Herz
//          rate             - bitrate in bytes
//          pack_size        - packet size in samples
//          channels         - number of channels; 1 for mono, 2 for stereo
void TestStereo::RegisterSendCodec(char side,
                                   char* codec_name,
                                   int32_t sampling_freq_hz,
                                   int rate,
                                   int pack_size,
                                   int channels) {}

void TestStereo::Run(TestPackStereo* channel,
                     int in_channels,
                     int out_channels,
                     int percent_loss) {}

void TestStereo::OpenOutFile(int16_t test_number) {}

}  // namespace webrtc