Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(28)

Side by Side Diff: webrtc/modules/audio_coding/main/test/TestStereo.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) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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 40 matching lines...) Expand 10 before | Expand all | Expand 10 after
51 const size_t payload_size, 51 const size_t payload_size,
52 const RTPFragmentationHeader* fragmentation) { 52 const RTPFragmentationHeader* fragmentation) {
53 WebRtcRTPHeader rtp_info; 53 WebRtcRTPHeader rtp_info;
54 int32_t status = 0; 54 int32_t status = 0;
55 55
56 rtp_info.header.markerBit = false; 56 rtp_info.header.markerBit = false;
57 rtp_info.header.ssrc = 0; 57 rtp_info.header.ssrc = 0;
58 rtp_info.header.sequenceNumber = seq_no_++; 58 rtp_info.header.sequenceNumber = seq_no_++;
59 rtp_info.header.payloadType = payload_type; 59 rtp_info.header.payloadType = payload_type;
60 rtp_info.header.timestamp = timestamp; 60 rtp_info.header.timestamp = timestamp;
61 if (frame_type == kFrameEmpty) { 61 if (frame_type == kEmptyFrame) {
62 // Skip this frame 62 // Skip this frame
63 return 0; 63 return 0;
64 } 64 }
65 65
66 if (lost_packet_ == false) { 66 if (lost_packet_ == false) {
67 if (frame_type != kAudioFrameCN) { 67 if (frame_type != kAudioFrameCN) {
68 rtp_info.type.Audio.isCNG = false; 68 rtp_info.type.Audio.isCNG = false;
69 rtp_info.type.Audio.channel = static_cast<int>(codec_mode_); 69 rtp_info.type.Audio.channel = static_cast<int>(codec_mode_);
70 } else { 70 } else {
71 rtp_info.type.Audio.isCNG = true; 71 rtp_info.type.Audio.isCNG = true;
(...skipping 756 matching lines...) Expand 10 before | Expand all | Expand 10 after
828 if (test_mode_ != 0) { 828 if (test_mode_ != 0) {
829 printf("%s -> ", my_codec_param.plname); 829 printf("%s -> ", my_codec_param.plname);
830 } 830 }
831 acm_b_->ReceiveCodec(&my_codec_param); 831 acm_b_->ReceiveCodec(&my_codec_param);
832 if (test_mode_ != 0) { 832 if (test_mode_ != 0) {
833 printf("%s\n", my_codec_param.plname); 833 printf("%s\n", my_codec_param.plname);
834 } 834 }
835 } 835 }
836 836
837 } // namespace webrtc 837 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/main/test/TestAllCodecs.cc ('k') | webrtc/modules/audio_coding/main/test/TestVADDTX.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698