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Diffstat (limited to 'webrtc/modules/audio_coding/main/test/TestStereo.h')
-rw-r--r-- | webrtc/modules/audio_coding/main/test/TestStereo.h | 117 |
1 files changed, 0 insertions, 117 deletions
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 b56e995272..0000000000 --- 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_ |