Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(168)

Side by Side Diff: webrtc/modules/audio_device/audio_transport.cc

Issue 2219653004: Remove old methods in AudioTransport, make it pass a gn build (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 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
(Empty)
1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
12
13 namespace webrtc {
14 // Some AudioTransport implementations do not care about these functions, so
15 // we provide default implementations.
16 void AudioTransport::PushCaptureData(int voe_channel,
17 const void* audio_data,
18 int bits_per_sample,
19 int sample_rate,
20 size_t number_of_channels,
21 size_t number_of_frames) {}
the sun 2016/08/05 18:14:31 I think this implementation file is unnecessary. E
Max Morin WebRTC 2016/08/08 15:28:57 Ok, they are pure virtual now.
22
23 void AudioTransport::PullRenderData(int bits_per_sample,
24 int sample_rate,
25 size_t number_of_channels,
26 size_t number_of_frames,
27 void* audio_data,
28 int64_t* elapsed_time_ms,
29 int64_t* ntp_time_ms) {}
30 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698