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

Issue 1334303005: Returning correct duration estimate on Opus DTX packets. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: fixing a silly mistake in unittest Created 5 years, 2 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>
(...skipping 87 matching lines...) Expand 10 before | Expand all | Expand 10 after
98 int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder, 98 int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
99 const int16_t* input_audio, 99 const int16_t* input_audio,
100 size_t input_samples, 100 size_t input_samples,
101 WebRtcOpusDecInst* decoder, 101 WebRtcOpusDecInst* decoder,
102 int16_t* output_audio, 102 int16_t* output_audio,
103 int16_t* audio_type) { 103 int16_t* audio_type) {
104 int encoded_bytes_int = WebRtcOpus_Encode(encoder, input_audio, input_samples, 104 int encoded_bytes_int = WebRtcOpus_Encode(encoder, input_audio, input_samples,
105 kMaxBytes, bitstream_); 105 kMaxBytes, bitstream_);
106 EXPECT_GE(encoded_bytes_int, 0); 106 EXPECT_GE(encoded_bytes_int, 0);
107 encoded_bytes_ = static_cast<size_t>(encoded_bytes_int); 107 encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
108 return WebRtcOpus_Decode(decoder, bitstream_, 108 int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
109 encoded_bytes_, output_audio, 109 int act_len = WebRtcOpus_Decode(decoder, bitstream_,
110 audio_type); 110 encoded_bytes_, output_audio,
111 audio_type);
112 EXPECT_EQ(est_len, act_len);
113 return act_len;
111 } 114 }
112 115
113 // Test if encoder/decoder can enter DTX mode properly and do not enter DTX when 116 // Test if encoder/decoder can enter DTX mode properly and do not enter DTX when
114 // they should not. This test is signal dependent. 117 // they should not. This test is signal dependent.
115 void OpusTest::TestDtxEffect(bool dtx) { 118 void OpusTest::TestDtxEffect(bool dtx) {
116 PrepareSpeechData(channels_, 20, 2000); 119 PrepareSpeechData(channels_, 20, 2000);
117 120
118 // Create encoder memory. 121 // Create encoder memory.
119 EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, 122 EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
120 channels_, 123 channels_,
(...skipping 493 matching lines...) Expand 10 before | Expand all | Expand 10 after
614 EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); 617 EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
615 EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); 618 EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
616 } 619 }
617 620
618 INSTANTIATE_TEST_CASE_P(VariousMode, 621 INSTANTIATE_TEST_CASE_P(VariousMode,
619 OpusTest, 622 OpusTest,
620 Combine(Values(1, 2), Values(0, 1))); 623 Combine(Values(1, 2), Values(0, 1)));
621 624
622 625
623 } // namespace webrtc 626 } // namespace webrtc
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