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Side by Side Diff: webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc

Issue 1319683002: AudioDecoder: Replace Init() with Reset() (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@buffer
Patch Set: review fixes Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 #include <string> 10 #include <string>
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369 369
370 // Encode & decode. 370 // Encode & decode.
371 int16_t audio_type; 371 int16_t audio_type;
372 int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_]; 372 int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
373 EXPECT_EQ(kOpus20msFrameSamples, 373 EXPECT_EQ(kOpus20msFrameSamples,
374 static_cast<size_t>(EncodeDecode( 374 static_cast<size_t>(EncodeDecode(
375 opus_encoder_, speech_data_.GetNextBlock(), 375 opus_encoder_, speech_data_.GetNextBlock(),
376 kOpus20msFrameSamples, opus_decoder_, output_data_decode, 376 kOpus20msFrameSamples, opus_decoder_, output_data_decode,
377 &audio_type))); 377 &audio_type)));
378 378
379 EXPECT_EQ(0, WebRtcOpus_DecoderInit(opus_decoder_)); 379 WebRtcOpus_DecoderInit(opus_decoder_);
380 380
381 EXPECT_EQ(kOpus20msFrameSamples, 381 EXPECT_EQ(kOpus20msFrameSamples,
382 static_cast<size_t>(WebRtcOpus_Decode( 382 static_cast<size_t>(WebRtcOpus_Decode(
383 opus_decoder_, bitstream_, encoded_bytes_, output_data_decode, 383 opus_decoder_, bitstream_, encoded_bytes_, output_data_decode,
384 &audio_type))); 384 &audio_type)));
385 385
386 // Free memory. 386 // Free memory.
387 delete[] output_data_decode; 387 delete[] output_data_decode;
388 EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); 388 EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
389 EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); 389 EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
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614 EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); 614 EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
615 EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); 615 EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
616 } 616 }
617 617
618 INSTANTIATE_TEST_CASE_P(VariousMode, 618 INSTANTIATE_TEST_CASE_P(VariousMode,
619 OpusTest, 619 OpusTest,
620 Combine(Values(1, 2), Values(0, 1))); 620 Combine(Values(1, 2), Values(0, 1)));
621 621
622 622
623 } // namespace webrtc 623 } // namespace webrtc
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