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7120742701
Now that we have moved WebRTC from src/webrtc to src/, common_types.h and typedefs.h are triggering a cpplint error. The cpplint complaint is: Include the directory when naming .h files [build/include] [4] This CL disables the error but we have to remove these two headers from the root directory. NOPRESUBMIT=true Bug: webrtc:5876 Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333 Reviewed-on: https://webrtc-review.googlesource.com/1577 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19859}
140 lines
4 KiB
C++
140 lines
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 MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
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#include <fstream>
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#include <memory>
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/audio_coding/neteq/include/neteq.h"
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#include "modules/audio_coding/neteq/tools/audio_sink.h"
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "modules/audio_coding/neteq/tools/rtp_generator.h"
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#include "modules/include/module_common_types.h"
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#include "rtc_base/flags.h"
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#include "test/gtest.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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namespace test {
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class LossModel {
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public:
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virtual ~LossModel() {};
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virtual bool Lost() = 0;
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};
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class NoLoss : public LossModel {
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public:
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bool Lost() override;
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};
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class UniformLoss : public LossModel {
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public:
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UniformLoss(double loss_rate);
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bool Lost() override;
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void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
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private:
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double loss_rate_;
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};
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class GilbertElliotLoss : public LossModel {
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public:
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GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
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~GilbertElliotLoss() override;
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bool Lost() override;
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private:
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// Prob. of losing current packet, when previous packet is lost.
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double prob_trans_11_;
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// Prob. of losing current packet, when previous packet is not lost.
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double prob_trans_01_;
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bool lost_last_;
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std::unique_ptr<UniformLoss> uniform_loss_model_;
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};
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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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NetEqDecoder decoder_type);
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~NetEqQualityTest() override;
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void SetUp() 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, size_t block_size_samples,
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rtc::Buffer* payload, size_t max_bytes) = 0;
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// PacketLost(...) determines weather a packet sent at an indicated time gets
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// lost or not.
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bool PacketLost();
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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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// Runs encoding / transmitting / decoding.
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void Simulate();
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// Write to log file. Usage Log() << ...
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std::ofstream& Log();
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NetEqDecoder decoder_type_;
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const size_t channels_;
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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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int packet_loss_rate_;
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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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// Number of samples per channel in a frame.
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const size_t in_size_samples_;
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size_t payload_size_bytes_;
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size_t max_payload_bytes_;
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std::unique_ptr<InputAudioFile> in_file_;
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std::unique_ptr<AudioSink> output_;
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std::ofstream log_file_;
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std::unique_ptr<RtpGenerator> rtp_generator_;
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std::unique_ptr<NetEq> neteq_;
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std::unique_ptr<LossModel> loss_model_;
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std::unique_ptr<int16_t[]> in_data_;
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rtc::Buffer payload_;
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AudioFrame out_frame_;
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RTPHeader rtp_header_;
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size_t total_payload_size_bytes_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
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