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_ |