Index: webrtc/modules/audio_coding/main/test/TestAllCodecs.h |
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h |
deleted file mode 100644 |
index 1cdc0cba980a9062d8549e4134d54cbe7e43ba2d..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h |
+++ /dev/null |
@@ -1,84 +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_TESTALLCODECS_H_ |
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_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" |
-#include "webrtc/typedefs.h" |
- |
-namespace webrtc { |
- |
-class Config; |
- |
-class TestPack : public AudioPacketizationCallback { |
- public: |
- TestPack(); |
- ~TestPack(); |
- |
- void RegisterReceiverACM(AudioCodingModule* acm); |
- |
- int32_t SendData(FrameType frame_type, |
- uint8_t payload_type, |
- uint32_t timestamp, |
- const uint8_t* payload_data, |
- size_t payload_size, |
- const RTPFragmentationHeader* fragmentation) override; |
- |
- size_t payload_size(); |
- uint32_t timestamp_diff(); |
- void reset_payload_size(); |
- |
- private: |
- AudioCodingModule* receiver_acm_; |
- uint16_t sequence_number_; |
- uint8_t payload_data_[60 * 32 * 2 * 2]; |
- uint32_t timestamp_diff_; |
- uint32_t last_in_timestamp_; |
- uint64_t total_bytes_; |
- size_t payload_size_; |
-}; |
- |
-class TestAllCodecs : public ACMTest { |
- public: |
- explicit TestAllCodecs(int test_mode); |
- ~TestAllCodecs(); |
- |
- void Perform() override; |
- |
- private: |
- // The default value of '-1' indicates that the registration is based only on |
- // codec name, and a sampling frequency matching is not required. |
- // This is useful for codecs which support several sampling frequency. |
- // Note! Only mono mode is tested in this test. |
- void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz, |
- int rate, int packet_size, size_t extra_byte); |
- |
- void Run(TestPack* channel); |
- void OpenOutFile(int test_number); |
- void DisplaySendReceiveCodec(); |
- |
- int test_mode_; |
- rtc::scoped_ptr<AudioCodingModule> acm_a_; |
- rtc::scoped_ptr<AudioCodingModule> acm_b_; |
- TestPack* channel_a_to_b_; |
- PCMFile infile_a_; |
- PCMFile outfile_b_; |
- int test_count_; |
- int packet_size_samples_; |
- size_t packet_size_bytes_; |
-}; |
- |
-} // namespace webrtc |
- |
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_ |