c8fa692ec4
I used a command like this to update the paths: perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"` The only manual edit is to add an include of webrtc/rtc_base/checks.h in webrtc/modules/audio_device/android/opensles_common.h, which likely was needed due to changed include paths due to 'git cl format'. BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2969653002 Cr-Commit-Position: refs/heads/master@{#18871}
97 lines
2.8 KiB
C++
97 lines
2.8 KiB
C++
/*
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* Copyright (c) 2014 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_ACM2_ACM_RECEIVE_TEST_H_
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#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
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#include <memory>
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#include <string>
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#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/scoped_ref_ptr.h"
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#include "webrtc/system_wrappers/include/clock.h"
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namespace webrtc {
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class AudioCodingModule;
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class AudioDecoder;
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struct CodecInst;
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namespace test {
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class AudioSink;
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class PacketSource;
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class AcmReceiveTestOldApi {
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public:
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enum NumOutputChannels {
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kArbitraryChannels = 0,
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kMonoOutput = 1,
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kStereoOutput = 2
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};
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AcmReceiveTestOldApi(PacketSource* packet_source,
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AudioSink* audio_sink,
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int output_freq_hz,
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NumOutputChannels exptected_output_channels,
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
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virtual ~AcmReceiveTestOldApi();
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// Registers the codecs with default parameters from ACM.
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void RegisterDefaultCodecs();
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// Registers codecs with payload types matching the pre-encoded NetEq test
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// files.
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void RegisterNetEqTestCodecs();
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// Runs the test and returns true if successful.
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void Run();
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AudioCodingModule* get_acm() { return acm_.get(); }
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protected:
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// Method is called after each block of output audio is received from ACM.
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virtual void AfterGetAudio() {}
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SimulatedClock clock_;
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std::unique_ptr<AudioCodingModule> acm_;
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PacketSource* packet_source_;
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AudioSink* audio_sink_;
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int output_freq_hz_;
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NumOutputChannels exptected_output_channels_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
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};
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// This test toggles the output frequency every |toggle_period_ms|. The test
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// starts with |output_freq_hz_1|. Except for the toggling, it does the same
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// thing as AcmReceiveTestOldApi.
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class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
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public:
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AcmReceiveTestToggleOutputFreqOldApi(
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PacketSource* packet_source,
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AudioSink* audio_sink,
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int output_freq_hz_1,
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int output_freq_hz_2,
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int toggle_period_ms,
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NumOutputChannels exptected_output_channels);
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protected:
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void AfterGetAudio() override;
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const int output_freq_hz_1_;
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const int output_freq_hz_2_;
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const int toggle_period_ms_;
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int64_t last_toggle_time_ms_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
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