c8e98187d1
This CL introduces a bit-exactness test for the receive-side of the AudioCoding Module. The main part of the test is done in the helper class AcmReceiveTest. The test is executed from the test fixture AcmReceiverBitExactness. The test inserts packets from a pre-encoded RTP file. The output is summed up into a checksum, which is verified versus a reference at the end of the test. Alternatively, if the flag --generate_output is given, the output is written to a file for subjective verification. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d
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/interface/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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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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virtual 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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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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