Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(164)

Side by Side Diff: webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.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
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 28 matching lines...) Expand all
39 39
40 } // namespace 40 } // namespace
41 41
42 class AcmReceiverTest : public AudioPacketizationCallback, 42 class AcmReceiverTest : public AudioPacketizationCallback,
43 public ::testing::Test { 43 public ::testing::Test {
44 protected: 44 protected:
45 AcmReceiverTest() 45 AcmReceiverTest()
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 AudioCoding::Config config; 50 AudioCoding::Config config;
51 config.transport = this; 51 config.transport = this;
52 acm_.reset(new AudioCodingImpl(config)); 52 acm_.reset(new AudioCodingImpl(config));
53 receiver_.reset(new AcmReceiver(config.ToOldConfig())); 53 receiver_.reset(new AcmReceiver(config.ToOldConfig()));
54 } 54 }
55 55
56 ~AcmReceiverTest() {} 56 ~AcmReceiverTest() {}
57 57
58 void SetUp() override { 58 void SetUp() override {
59 ASSERT_TRUE(receiver_.get() != NULL); 59 ASSERT_TRUE(receiver_.get() != NULL);
(...skipping 54 matching lines...) Expand 10 before | Expand all | Expand 10 after
114 ++n; 114 ++n;
115 } 115 }
116 } 116 }
117 117
118 int32_t SendData(FrameType frame_type, 118 int32_t SendData(FrameType frame_type,
119 uint8_t payload_type, 119 uint8_t payload_type,
120 uint32_t timestamp, 120 uint32_t timestamp,
121 const uint8_t* payload_data, 121 const uint8_t* payload_data,
122 size_t payload_len_bytes, 122 size_t payload_len_bytes,
123 const RTPFragmentationHeader* fragmentation) override { 123 const RTPFragmentationHeader* fragmentation) override {
124 if (frame_type == kFrameEmpty) 124 if (frame_type == kEmptyFrame)
125 return 0; 125 return 0;
126 126
127 rtp_header_.header.payloadType = payload_type; 127 rtp_header_.header.payloadType = payload_type;
128 rtp_header_.frameType = frame_type; 128 rtp_header_.frameType = frame_type;
129 if (frame_type == kAudioFrameSpeech) 129 if (frame_type == kAudioFrameSpeech)
130 rtp_header_.type.Audio.isCNG = false; 130 rtp_header_.type.Audio.isCNG = false;
131 else 131 else
132 rtp_header_.type.Audio.isCNG = true; 132 rtp_header_.type.Audio.isCNG = true;
133 rtp_header_.header.timestamp = timestamp; 133 rtp_header_.header.timestamp = timestamp;
134 134
(...skipping 226 matching lines...) Expand 10 before | Expand all | Expand 10 after
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
OLDNEW
« no previous file with comments | « webrtc/frame_callback.h ('k') | webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698