Files
webrtc/webrtc/modules/audio_coding/main/acm2/acm_receive_test.h
henrik.lundin@webrtc.org 0e6e4d2ff2 Reland "Converting five tests to use new AudioCoding interface" (r7258)
This CL reverts r7264. The problem was that iSAC-SWB and iSAC-FB are
not supported on android. These are now disabled.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7273 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 12:05:34 +00:00

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1.7 KiB
C++

/*
* Copyright (c) 2014 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_ACM2_ACM_RECEIVE_TEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class AudioCoding;
struct CodecInst;
namespace test {
class AudioSink;
class PacketSource;
class AcmReceiveTest {
public:
enum NumOutputChannels {
kArbitraryChannels = 0,
kMonoOutput = 1,
kStereoOutput = 2
};
AcmReceiveTest(
PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz,
NumOutputChannels exptected_output_channels);
virtual ~AcmReceiveTest() {}
// Registers the codecs with default parameters from ACM.
void RegisterDefaultCodecs();
// Registers codecs with payload types matching the pre-encoded NetEq test
// files.
void RegisterNetEqTestCodecs();
// Runs the test and returns true if successful.
void Run();
private:
SimulatedClock clock_;
scoped_ptr<AudioCoding> acm_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
const int output_freq_hz_;
NumOutputChannels exptected_output_channels_;
DISALLOW_COPY_AND_ASSIGN(AcmReceiveTest);
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_