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
116 lines
4.1 KiB
C++
116 lines
4.1 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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#include <stdio.h>
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#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
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namespace webrtc {
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namespace test {
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const uint8_t kPayloadType = 95;
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const int kOutputSizeMs = 10;
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NetEqQualityTest::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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: decoded_time_ms_(0),
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decodable_time_ms_(0),
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drift_factor_(drift_factor),
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block_duration_ms_(block_duration_ms),
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in_sampling_khz_(in_sampling_khz),
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out_sampling_khz_(out_sampling_khz),
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decoder_type_(decoder_type),
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channels_(channels),
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in_filename_(in_filename),
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out_filename_(out_filename),
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in_size_samples_(in_sampling_khz_ * block_duration_ms_),
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out_size_samples_(out_sampling_khz_ * kOutputSizeMs),
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payload_size_bytes_(0),
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max_payload_bytes_(0),
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in_file_(new InputAudioFile(in_filename_)),
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out_file_(NULL),
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rtp_generator_(new RtpGenerator(in_sampling_khz_, 0, 0,
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decodable_time_ms_)) {
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NetEq::Config config;
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config.sample_rate_hz = out_sampling_khz_ * 1000;
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neteq_.reset(NetEq::Create(config));
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max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
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in_data_.reset(new int16_t[in_size_samples_ * channels_]);
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payload_.reset(new uint8_t[max_payload_bytes_]);
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out_data_.reset(new int16_t[out_size_samples_ * channels_]);
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}
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void NetEqQualityTest::SetUp() {
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out_file_ = fopen(out_filename_.c_str(), "wb");
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ASSERT_TRUE(out_file_ != NULL);
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ASSERT_EQ(0, neteq_->RegisterPayloadType(decoder_type_, kPayloadType));
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rtp_generator_->set_drift_factor(drift_factor_);
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}
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void NetEqQualityTest::TearDown() {
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fclose(out_file_);
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}
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int NetEqQualityTest::Transmit() {
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int packet_input_time_ms =
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rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_,
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&rtp_header_);
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if (!PacketLost(packet_input_time_ms) && payload_size_bytes_ > 0) {
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int ret = neteq_->InsertPacket(rtp_header_, &payload_[0],
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payload_size_bytes_,
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packet_input_time_ms * in_sampling_khz_);
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if (ret != NetEq::kOK)
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return -1;
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}
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return packet_input_time_ms;
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}
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int NetEqQualityTest::DecodeBlock() {
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int channels;
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int samples;
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int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
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&samples, &channels, NULL);
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if (ret != NetEq::kOK) {
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return -1;
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} else {
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assert(channels == channels_);
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assert(samples == kOutputSizeMs * out_sampling_khz_);
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fwrite(&out_data_[0], sizeof(int16_t), samples * channels, out_file_);
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return samples;
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}
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}
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void NetEqQualityTest::Simulate(int end_time_ms) {
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int audio_size_samples;
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while (decoded_time_ms_ < end_time_ms) {
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while (decodable_time_ms_ - kOutputSizeMs < decoded_time_ms_) {
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ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0]));
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payload_size_bytes_ = EncodeBlock(&in_data_[0],
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in_size_samples_, &payload_[0],
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max_payload_bytes_);
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decodable_time_ms_ = Transmit() + block_duration_ms_;
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}
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audio_size_samples = DecodeBlock();
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if (audio_size_samples > 0) {
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decoded_time_ms_ += audio_size_samples / out_sampling_khz_;
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}
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}
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}
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} // namespace test
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} // namespace webrtc
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