9c55f0f957
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for a hard-coded reference to change. This CL effectively reverts r6257 "Rename neteq4 folder to neteq". BUG=2996 TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
101 lines
3.4 KiB
C++
101 lines
3.4 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_NETEQ_QUALITY_TEST_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
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#include <string>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
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#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.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 NetEqQualityTest : public ::testing::Test {
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protected:
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NetEqQualityTest(int block_duration_ms,
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int in_sampling_khz,
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int out_sampling_khz,
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enum NetEqDecoder decoder_type,
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int channels,
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double drift_factor,
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std::string in_filename,
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std::string out_filename);
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virtual void SetUp() OVERRIDE;
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virtual void TearDown() OVERRIDE;
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// EncodeBlock(...) does the following:
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// 1. encodes a block of audio, saved in |in_data| and has a length of
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// |block_size_samples| (samples per channel),
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// 2. save the bit stream to |payload| of |max_bytes| bytes in size,
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// 3. returns the length of the payload (in bytes),
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virtual int EncodeBlock(int16_t* in_data, int block_size_samples,
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uint8_t* payload, int max_bytes) = 0;
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// PacketLoss(...) determines weather a packet sent at an indicated time gets
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// lost or not.
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virtual bool PacketLost(int packet_input_time_ms) { return false; }
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// DecodeBlock() decodes a block of audio using the payload stored in
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// |payload_| with the length of |payload_size_bytes_| (bytes). The decoded
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// audio is to be stored in |out_data_|.
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int DecodeBlock();
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// Transmit() uses |rtp_generator_| to generate a packet and passes it to
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// |neteq_|.
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int Transmit();
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// Simulate(...) runs encoding / transmitting / decoding up to |end_time_ms|
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// (miliseconds), the resulted audio is stored in the file with the name of
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// |out_filename_|.
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void Simulate(int end_time_ms);
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private:
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int decoded_time_ms_;
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int decodable_time_ms_;
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double drift_factor_;
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const int block_duration_ms_;
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const int in_sampling_khz_;
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const int out_sampling_khz_;
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const enum NetEqDecoder decoder_type_;
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const int channels_;
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const std::string in_filename_;
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const std::string out_filename_;
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// Number of samples per channel in a frame.
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const int in_size_samples_;
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// Expected output number of samples per channel in a frame.
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const int out_size_samples_;
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int payload_size_bytes_;
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int max_payload_bytes_;
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scoped_ptr<InputAudioFile> in_file_;
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FILE* out_file_;
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scoped_ptr<RtpGenerator> rtp_generator_;
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scoped_ptr<NetEq> neteq_;
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scoped_ptr<int16_t[]> in_data_;
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scoped_ptr<uint8_t[]> payload_;
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scoped_ptr<int16_t[]> out_data_;
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WebRtcRTPHeader rtp_header_;
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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_NETEQ_QUALITY_TEST_H_
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