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Side by Side Diff: webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc

Issue 1371043003: Unify FrameType and VideoFrameType. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase 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 10
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39 39
40 } // namespace 40 } // namespace
41 41
42 class AcmReceiverTestOldApi : public AudioPacketizationCallback, 42 class AcmReceiverTestOldApi : public AudioPacketizationCallback,
43 public ::testing::Test { 43 public ::testing::Test {
44 protected: 44 protected:
45 AcmReceiverTestOldApi() 45 AcmReceiverTestOldApi()
46 : timestamp_(0), 46 : timestamp_(0),
47 packet_sent_(false), 47 packet_sent_(false),
48 last_packet_send_timestamp_(timestamp_), 48 last_packet_send_timestamp_(timestamp_),
49 last_frame_type_(kFrameEmpty) { 49 last_frame_type_(kEmptyFrame) {
50 AudioCodingModule::Config config; 50 AudioCodingModule::Config config;
51 acm_.reset(new AudioCodingModuleImpl(config)); 51 acm_.reset(new AudioCodingModuleImpl(config));
52 receiver_.reset(new AcmReceiver(config)); 52 receiver_.reset(new AcmReceiver(config));
53 } 53 }
54 54
55 ~AcmReceiverTestOldApi() {} 55 ~AcmReceiverTestOldApi() {}
56 56
57 void SetUp() override { 57 void SetUp() override {
58 ASSERT_TRUE(receiver_.get() != NULL); 58 ASSERT_TRUE(receiver_.get() != NULL);
59 ASSERT_TRUE(acm_.get() != NULL); 59 ASSERT_TRUE(acm_.get() != NULL);
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113 ++n; 113 ++n;
114 } 114 }
115 } 115 }
116 116
117 int SendData(FrameType frame_type, 117 int SendData(FrameType frame_type,
118 uint8_t payload_type, 118 uint8_t payload_type,
119 uint32_t timestamp, 119 uint32_t timestamp,
120 const uint8_t* payload_data, 120 const uint8_t* payload_data,
121 size_t payload_len_bytes, 121 size_t payload_len_bytes,
122 const RTPFragmentationHeader* fragmentation) override { 122 const RTPFragmentationHeader* fragmentation) override {
123 if (frame_type == kFrameEmpty) 123 if (frame_type == kEmptyFrame)
124 return 0; 124 return 0;
125 125
126 rtp_header_.header.payloadType = payload_type; 126 rtp_header_.header.payloadType = payload_type;
127 rtp_header_.frameType = frame_type; 127 rtp_header_.frameType = frame_type;
128 if (frame_type == kAudioFrameSpeech) 128 if (frame_type == kAudioFrameSpeech)
129 rtp_header_.type.Audio.isCNG = false; 129 rtp_header_.type.Audio.isCNG = false;
130 else 130 else
131 rtp_header_.type.Audio.isCNG = true; 131 rtp_header_.type.Audio.isCNG = true;
132 rtp_header_.header.timestamp = timestamp; 132 rtp_header_.header.timestamp = timestamp;
133 133
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361 EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id()); 361 EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id());
362 EXPECT_EQ(0, receiver_->LastAudioCodec(&codec)); 362 EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
363 EXPECT_TRUE(CodecsEqual(codecs_[kCodecId[n]], codec)); 363 EXPECT_TRUE(CodecsEqual(codecs_[kCodecId[n]], codec));
364 ++n; 364 ++n;
365 } 365 }
366 } 366 }
367 367
368 } // namespace acm2 368 } // namespace acm2
369 369
370 } // namespace webrtc 370 } // namespace webrtc
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