ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
64 lines
1.9 KiB
C++
64 lines
1.9 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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namespace test {
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// Interface class for an object receiving raw output audio from test
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// applications.
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class AudioSink {
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public:
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AudioSink() {}
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virtual ~AudioSink() {}
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// Writes |num_samples| from |audio| to the AudioSink. Returns true if
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// successful, otherwise false.
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virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
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// Writes |audio_frame| to the AudioSink. Returns true if successful,
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// otherwise false.
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bool WriteAudioFrame(const AudioFrame& audio_frame) {
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return WriteArray(
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audio_frame.data_,
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audio_frame.samples_per_channel_ * audio_frame.num_channels_);
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}
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioSink);
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};
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// Forks the output audio to two AudioSink objects.
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class AudioSinkFork : public AudioSink {
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public:
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AudioSinkFork(AudioSink* left, AudioSink* right)
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: left_sink_(left), right_sink_(right) {}
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bool WriteArray(const int16_t* audio, size_t num_samples) override {
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return left_sink_->WriteArray(audio, num_samples) &&
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right_sink_->WriteArray(audio, num_samples);
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}
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private:
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AudioSink* left_sink_;
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AudioSink* right_sink_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
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