36b1a5fcec
const int16_t* data() const; int16_t* mutable_data(); - data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames. - mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_. These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation. This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later. BUG=webrtc:7343 TBR=henrika Review-Url: https://codereview.webrtc.org/2750783004 Cr-Commit-Position: refs/heads/master@{#18543}
72 lines
2.1 KiB
C++
72 lines
2.1 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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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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// An AudioSink implementation that does nothing.
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class VoidAudioSink : public AudioSink {
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public:
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VoidAudioSink() = default;
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bool WriteArray(const int16_t* audio, size_t num_samples) override;
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(VoidAudioSink);
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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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