blob: 6349c6392a8c68e1b7cc59838d71787de91421e9 [file] [log] [blame]
* 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.
#include <stddef.h> // for size_t
#include <memory>
#include <string>
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/scoped_refptr.h"
#include "rtc_base/constructor_magic.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
class AudioCodingModule;
class AudioDecoder;
namespace test {
class AudioSink;
class PacketSource;
class AcmReceiveTestOldApi {
enum NumOutputChannels : size_t {
kArbitraryChannels = 0,
kMonoOutput = 1,
kStereoOutput = 2,
kQuadOutput = 4
AcmReceiveTestOldApi(PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz,
NumOutputChannels exptected_output_channels,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
virtual ~AcmReceiveTestOldApi();
// 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();
AudioCodingModule* get_acm() { return acm_.get(); }
// Method is called after each block of output audio is received from ACM.
virtual void AfterGetAudio() {}
SimulatedClock clock_;
std::unique_ptr<AudioCodingModule> acm_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
int output_freq_hz_;
NumOutputChannels exptected_output_channels_;
// This test toggles the output frequency every `toggle_period_ms`. The test
// starts with `output_freq_hz_1`. Except for the toggling, it does the same
// thing as AcmReceiveTestOldApi.
class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz_1,
int output_freq_hz_2,
int toggle_period_ms,
NumOutputChannels exptected_output_channels);
void AfterGetAudio() override;
const int output_freq_hz_1_;
const int output_freq_hz_2_;
const int toggle_period_ms_;
int64_t last_toggle_time_ms_;
} // namespace test
} // namespace webrtc