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Unified Diff: webrtc/modules/audio_coding/main/test/TestStereo.h

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years, 1 month ago
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Index: webrtc/modules/audio_coding/main/test/TestStereo.h
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.h b/webrtc/modules/audio_coding/main/test/TestStereo.h
deleted file mode 100644
index b56e9952724824829eaf652a2d4170fbf7515b78..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/TestStereo.h
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
-
-#include <math.h>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-
-#define PCMA_AND_PCMU
-
-namespace webrtc {
-
-enum StereoMonoMode {
- kNotSet,
- kMono,
- kStereo
-};
-
-class TestPackStereo : public AudioPacketizationCallback {
- public:
- TestPackStereo();
- ~TestPackStereo();
-
- void RegisterReceiverACM(AudioCodingModule* acm);
-
- int32_t SendData(const FrameType frame_type,
- const uint8_t payload_type,
- const uint32_t timestamp,
- const uint8_t* payload_data,
- const size_t payload_size,
- const RTPFragmentationHeader* fragmentation) override;
-
- uint16_t payload_size();
- uint32_t timestamp_diff();
- void reset_payload_size();
- void set_codec_mode(StereoMonoMode mode);
- void set_lost_packet(bool lost);
-
- private:
- AudioCodingModule* receiver_acm_;
- int16_t seq_no_;
- uint32_t timestamp_diff_;
- uint32_t last_in_timestamp_;
- uint64_t total_bytes_;
- int payload_size_;
- StereoMonoMode codec_mode_;
- // Simulate packet losses
- bool lost_packet_;
-};
-
-class TestStereo : public ACMTest {
- public:
- explicit TestStereo(int test_mode);
- ~TestStereo();
-
- void Perform() override;
-
- private:
- // The default value of '-1' indicates that the registration is based only on
- // codec name and a sampling frequncy matching is not required. This is useful
- // for codecs which support several sampling frequency.
- void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz,
- int rate, int pack_size, int channels,
- int payload_type);
-
- void Run(TestPackStereo* channel, int in_channels, int out_channels,
- int percent_loss = 0);
- void OpenOutFile(int16_t test_number);
- void DisplaySendReceiveCodec();
-
- int test_mode_;
-
- rtc::scoped_ptr<AudioCodingModule> acm_a_;
- rtc::scoped_ptr<AudioCodingModule> acm_b_;
-
- TestPackStereo* channel_a2b_;
-
- PCMFile* in_file_stereo_;
- PCMFile* in_file_mono_;
- PCMFile out_file_;
- int16_t test_cntr_;
- uint16_t pack_size_samp_;
- uint16_t pack_size_bytes_;
- int counter_;
- char* send_codec_name_;
-
- // Payload types for stereo codecs and CNG
-#ifdef WEBRTC_CODEC_G722
- int g722_pltype_;
-#endif
- int l16_8khz_pltype_;
- int l16_16khz_pltype_;
- int l16_32khz_pltype_;
-#ifdef PCMA_AND_PCMU
- int pcma_pltype_;
- int pcmu_pltype_;
-#endif
-#ifdef WEBRTC_CODEC_OPUS
- int opus_pltype_;
-#endif
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
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