Adding test::AudioSink interface and derived classes
The AudioSink interface is supposed to be used by tests that produce audio output. Two implementation classes are also provided: OutputAudioFile: Writes the audio to a pcm file. AudioChecksum: Calculates the MD5 checksum of the audio. These will both be used in future changes. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6490 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -182,10 +182,13 @@
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'tools',
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'tools',
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],
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],
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'sources': [
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'sources': [
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'tools/audio_checksum.h',
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'tools/audio_loop.cc',
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'tools/audio_loop.cc',
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'tools/audio_loop.h',
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'tools/audio_loop.h',
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'tools/audio_sink.h',
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'tools/input_audio_file.cc',
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'tools/input_audio_file.cc',
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'tools/input_audio_file.h',
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'tools/input_audio_file.h',
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'tools/output_audio_file.h',
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'tools/packet.cc',
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'tools/packet.cc',
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'tools/packet.h',
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'tools/packet.h',
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'tools/packet_source.h',
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'tools/packet_source.h',
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60
webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
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webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
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/*
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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_CHECKSUM_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
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#include <string>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/md5digest.h"
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#include "webrtc/base/stringencode.h"
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#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
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#include "webrtc/system_wrappers/interface/compile_assert.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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class AudioChecksum : public AudioSink {
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public:
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AudioChecksum() : finished_(false) {}
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virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
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if (finished_)
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return false;
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#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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#error "Big-endian gives a different checksum"
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#endif
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checksum_.Update(audio, num_samples * sizeof(*audio));
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return true;
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}
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// Finalizes the computations, and returns the checksum.
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std::string Finish() {
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if (!finished_) {
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finished_ = true;
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checksum_.Finish(checksum_result_, rtc::Md5Digest::kSize);
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}
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return rtc::hex_encode(checksum_result_, rtc::Md5Digest::kSize);
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}
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private:
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rtc::Md5Digest checksum_;
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char checksum_result_[rtc::Md5Digest::kSize];
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bool finished_;
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DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
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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_CHECKSUM_H_
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46
webrtc/modules/audio_coding/neteq/tools/audio_sink.h
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webrtc/modules/audio_coding/neteq/tools/audio_sink.h
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/*
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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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} // 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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50
webrtc/modules/audio_coding/neteq/tools/output_audio_file.h
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webrtc/modules/audio_coding/neteq/tools/output_audio_file.h
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/*
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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_OUTPUT_AUDIO_FILE_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
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#include <assert.h>
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#include <stdio.h>
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#include <string>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
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namespace webrtc {
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namespace test {
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class OutputAudioFile : public AudioSink {
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public:
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// Creates an OutputAudioFile, opening a file named |file_name| for writing.
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// The file format is 16-bit signed host-endian PCM.
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explicit OutputAudioFile(const std::string& file_name) {
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out_file_ = fopen(file_name.c_str(), "wb");
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}
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virtual ~OutputAudioFile() {
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if (out_file_)
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fclose(out_file_);
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}
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virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
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assert(out_file_);
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return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
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}
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private:
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FILE* out_file_;
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DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
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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_OUTPUT_AUDIO_FILE_H_
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