Index: webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h |
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h b/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h |
deleted file mode 100644 |
index 0b5671fe8c538af10c514f90a9315f7892655069..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h |
+++ /dev/null |
@@ -1,94 +0,0 @@ |
-/* |
- * 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/base/scoped_ptr.h" |
-#include "webrtc/system_wrappers/include/clock.h" |
- |
-namespace webrtc { |
-class AudioCodingModule; |
-class AudioDecoder; |
-struct CodecInst; |
- |
-namespace test { |
-class AudioSink; |
-class PacketSource; |
- |
-class AcmReceiveTestOldApi { |
- public: |
- enum NumOutputChannels { |
- kArbitraryChannels = 0, |
- kMonoOutput = 1, |
- kStereoOutput = 2 |
- }; |
- |
- AcmReceiveTestOldApi(PacketSource* packet_source, |
- AudioSink* audio_sink, |
- int output_freq_hz, |
- NumOutputChannels exptected_output_channels); |
- 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(); |
- |
- int RegisterExternalReceiveCodec(int rtp_payload_type, |
- AudioDecoder* external_decoder, |
- int sample_rate_hz, |
- int num_channels); |
- |
- // Runs the test and returns true if successful. |
- void Run(); |
- |
- protected: |
- // Method is called after each block of output audio is received from ACM. |
- virtual void AfterGetAudio() {} |
- |
- SimulatedClock clock_; |
- rtc::scoped_ptr<AudioCodingModule> acm_; |
- PacketSource* packet_source_; |
- AudioSink* audio_sink_; |
- int output_freq_hz_; |
- NumOutputChannels exptected_output_channels_; |
- |
- RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi); |
-}; |
- |
-// 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 { |
- public: |
- AcmReceiveTestToggleOutputFreqOldApi( |
- PacketSource* packet_source, |
- AudioSink* audio_sink, |
- int output_freq_hz_1, |
- int output_freq_hz_2, |
- int toggle_period_ms, |
- NumOutputChannels exptected_output_channels); |
- |
- protected: |
- 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 |
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_ |