Rename neteq4 folder to neteq
BUG=2996 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12569005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
931
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
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931
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -0,0 +1,931 @@
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/*
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* Copyright (c) 2012 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 "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
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#include <assert.h>
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#include <stdlib.h>
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#include <string>
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#include "gtest/gtest.h"
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#ifdef WEBRTC_CODEC_CELT
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#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h"
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#endif
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#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
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#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
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#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
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#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
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#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
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#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
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#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
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#include "webrtc/system_wrappers/interface/data_log.h"
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#include "webrtc/test/testsupport/fileutils.h"
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namespace webrtc {
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class AudioDecoderTest : public ::testing::Test {
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protected:
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AudioDecoderTest()
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: input_fp_(NULL),
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input_(NULL),
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encoded_(NULL),
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decoded_(NULL),
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frame_size_(0),
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data_length_(0),
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encoded_bytes_(0),
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channels_(1),
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decoder_(NULL) {
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input_file_ = webrtc::test::ProjectRootPath() +
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"resources/audio_coding/testfile32kHz.pcm";
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}
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virtual ~AudioDecoderTest() {}
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virtual void SetUp() {
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// Create arrays.
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ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0";
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input_ = new int16_t[data_length_];
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encoded_ = new uint8_t[data_length_ * 2];
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decoded_ = new int16_t[data_length_ * channels_];
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// Open input file.
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input_fp_ = fopen(input_file_.c_str(), "rb");
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ASSERT_TRUE(input_fp_ != NULL) << "Failed to open file " << input_file_;
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// Read data to |input_|.
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ASSERT_EQ(data_length_,
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fread(input_, sizeof(int16_t), data_length_, input_fp_)) <<
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"Could not read enough data from file";
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// Logging to view input and output in Matlab.
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// Use 'gyp -Denable_data_logging=1' to enable logging.
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DataLog::CreateLog();
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DataLog::AddTable("CodecTest");
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DataLog::AddColumn("CodecTest", "input", 1);
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DataLog::AddColumn("CodecTest", "output", 1);
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}
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virtual void TearDown() {
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delete decoder_;
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decoder_ = NULL;
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// Close input file.
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fclose(input_fp_);
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// Delete arrays.
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delete [] input_;
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input_ = NULL;
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delete [] encoded_;
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encoded_ = NULL;
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delete [] decoded_;
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decoded_ = NULL;
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// Close log.
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DataLog::ReturnLog();
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}
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virtual void InitEncoder() { }
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// This method must be implemented for all tests derived from this class.
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virtual int EncodeFrame(const int16_t* input, size_t input_len,
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uint8_t* output) = 0;
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// Encodes and decodes audio. The absolute difference between the input and
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// output is compared vs |tolerance|, and the mean-squared error is compared
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// with |mse|. The encoded stream should contain |expected_bytes|. For stereo
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// audio, the absolute difference between the two channels is compared vs
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// |channel_diff_tolerance|.
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void EncodeDecodeTest(size_t expected_bytes, int tolerance, double mse,
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int delay = 0, int channel_diff_tolerance = 0) {
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ASSERT_GE(tolerance, 0) << "Test must define a tolerance >= 0";
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ASSERT_GE(channel_diff_tolerance, 0) <<
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"Test must define a channel_diff_tolerance >= 0";
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size_t processed_samples = 0u;
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encoded_bytes_ = 0u;
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InitEncoder();
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EXPECT_EQ(0, decoder_->Init());
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while (processed_samples + frame_size_ <= data_length_) {
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size_t enc_len = EncodeFrame(&input_[processed_samples], frame_size_,
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&encoded_[encoded_bytes_]);
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AudioDecoder::SpeechType speech_type;
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size_t dec_len = decoder_->Decode(&encoded_[encoded_bytes_], enc_len,
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&decoded_[processed_samples *
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channels_],
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&speech_type);
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EXPECT_EQ(frame_size_ * channels_, dec_len);
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encoded_bytes_ += enc_len;
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processed_samples += frame_size_;
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}
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// For some codecs it doesn't make sense to check expected number of bytes,
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// since the number can vary for different platforms. Opus and iSAC are
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// such codecs. In this case expected_bytes is set to 0.
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if (expected_bytes) {
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EXPECT_EQ(expected_bytes, encoded_bytes_);
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}
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CompareInputOutput(processed_samples, tolerance, delay);
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if (channels_ == 2)
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CompareTwoChannels(processed_samples, channel_diff_tolerance);
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EXPECT_LE(MseInputOutput(processed_samples, delay), mse);
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}
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// The absolute difference between the input and output (the first channel) is
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// compared vs |tolerance|. The parameter |delay| is used to correct for codec
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// delays.
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virtual void CompareInputOutput(size_t num_samples, int tolerance,
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int delay) const {
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assert(num_samples <= data_length_);
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for (unsigned int n = 0; n < num_samples - delay; ++n) {
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ASSERT_NEAR(input_[n], decoded_[channels_ * n + delay], tolerance) <<
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"Exit test on first diff; n = " << n;
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DataLog::InsertCell("CodecTest", "input", input_[n]);
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DataLog::InsertCell("CodecTest", "output", decoded_[channels_ * n]);
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DataLog::NextRow("CodecTest");
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}
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}
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// The absolute difference between the two channels in a stereo is compared vs
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// |tolerance|.
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virtual void CompareTwoChannels(size_t samples_per_channel,
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int tolerance) const {
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assert(samples_per_channel <= data_length_);
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for (unsigned int n = 0; n < samples_per_channel; ++n)
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ASSERT_NEAR(decoded_[channels_ * n], decoded_[channels_ * n + 1],
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tolerance) << "Stereo samples differ.";
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}
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// Calculates mean-squared error between input and output (the first channel).
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// The parameter |delay| is used to correct for codec delays.
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virtual double MseInputOutput(size_t num_samples, int delay) const {
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assert(num_samples <= data_length_);
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if (num_samples == 0) return 0.0;
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double squared_sum = 0.0;
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for (unsigned int n = 0; n < num_samples - delay; ++n) {
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squared_sum += (input_[n] - decoded_[channels_ * n + delay]) *
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(input_[n] - decoded_[channels_ * n + delay]);
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}
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return squared_sum / (num_samples - delay);
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}
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// Encodes a payload and decodes it twice with decoder re-init before each
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// decode. Verifies that the decoded result is the same.
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void ReInitTest() {
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uint8_t* encoded = encoded_;
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uint8_t* encoded_copy = encoded_ + 2 * frame_size_;
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int16_t* output1 = decoded_;
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int16_t* output2 = decoded_ + frame_size_;
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InitEncoder();
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size_t enc_len = EncodeFrame(input_, frame_size_, encoded);
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size_t dec_len;
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// Copy payload since iSAC fix destroys it during decode.
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// Issue: http://code.google.com/p/webrtc/issues/detail?id=845.
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// TODO(hlundin): Remove if the iSAC bug gets fixed.
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memcpy(encoded_copy, encoded, enc_len);
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AudioDecoder::SpeechType speech_type1, speech_type2;
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EXPECT_EQ(0, decoder_->Init());
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dec_len = decoder_->Decode(encoded, enc_len, output1, &speech_type1);
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EXPECT_EQ(frame_size_ * channels_, dec_len);
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// Re-init decoder and decode again.
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EXPECT_EQ(0, decoder_->Init());
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dec_len = decoder_->Decode(encoded_copy, enc_len, output2, &speech_type2);
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EXPECT_EQ(frame_size_ * channels_, dec_len);
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for (unsigned int n = 0; n < frame_size_; ++n) {
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ASSERT_EQ(output1[n], output2[n]) << "Exit test on first diff; n = " << n;
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}
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EXPECT_EQ(speech_type1, speech_type2);
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}
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// Call DecodePlc and verify that the correct number of samples is produced.
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void DecodePlcTest() {
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InitEncoder();
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size_t enc_len = EncodeFrame(input_, frame_size_, encoded_);
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AudioDecoder::SpeechType speech_type;
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EXPECT_EQ(0, decoder_->Init());
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size_t dec_len =
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decoder_->Decode(encoded_, enc_len, decoded_, &speech_type);
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EXPECT_EQ(frame_size_ * channels_, dec_len);
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// Call DecodePlc and verify that we get one frame of data.
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// (Overwrite the output from the above Decode call, but that does not
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// matter.)
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dec_len = decoder_->DecodePlc(1, decoded_);
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EXPECT_EQ(frame_size_ * channels_, dec_len);
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}
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std::string input_file_;
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FILE* input_fp_;
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int16_t* input_;
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uint8_t* encoded_;
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int16_t* decoded_;
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size_t frame_size_;
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size_t data_length_;
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size_t encoded_bytes_;
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size_t channels_;
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AudioDecoder* decoder_;
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};
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class AudioDecoderPcmUTest : public AudioDecoderTest {
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protected:
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AudioDecoderPcmUTest() : AudioDecoderTest() {
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frame_size_ = 160;
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data_length_ = 10 * frame_size_;
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decoder_ = new AudioDecoderPcmU;
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assert(decoder_);
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}
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virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
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uint8_t* output) {
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int enc_len_bytes =
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WebRtcG711_EncodeU(NULL, const_cast<int16_t*>(input),
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static_cast<int>(input_len_samples),
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reinterpret_cast<int16_t*>(output));
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EXPECT_EQ(input_len_samples, static_cast<size_t>(enc_len_bytes));
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return enc_len_bytes;
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}
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};
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class AudioDecoderPcmATest : public AudioDecoderTest {
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protected:
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AudioDecoderPcmATest() : AudioDecoderTest() {
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frame_size_ = 160;
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data_length_ = 10 * frame_size_;
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decoder_ = new AudioDecoderPcmA;
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assert(decoder_);
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}
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virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
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uint8_t* output) {
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int enc_len_bytes =
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WebRtcG711_EncodeA(NULL, const_cast<int16_t*>(input),
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static_cast<int>(input_len_samples),
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reinterpret_cast<int16_t*>(output));
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EXPECT_EQ(input_len_samples, static_cast<size_t>(enc_len_bytes));
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return enc_len_bytes;
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}
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};
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class AudioDecoderPcm16BTest : public AudioDecoderTest {
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protected:
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AudioDecoderPcm16BTest() : AudioDecoderTest() {
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frame_size_ = 160;
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data_length_ = 10 * frame_size_;
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decoder_ = new AudioDecoderPcm16B(kDecoderPCM16B);
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assert(decoder_);
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}
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virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
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uint8_t* output) {
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int enc_len_bytes = WebRtcPcm16b_EncodeW16(
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const_cast<int16_t*>(input), static_cast<int>(input_len_samples),
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reinterpret_cast<int16_t*>(output));
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EXPECT_EQ(2 * input_len_samples, static_cast<size_t>(enc_len_bytes));
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return enc_len_bytes;
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}
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};
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class AudioDecoderIlbcTest : public AudioDecoderTest {
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protected:
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AudioDecoderIlbcTest() : AudioDecoderTest() {
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frame_size_ = 240;
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data_length_ = 10 * frame_size_;
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decoder_ = new AudioDecoderIlbc;
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assert(decoder_);
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WebRtcIlbcfix_EncoderCreate(&encoder_);
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}
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~AudioDecoderIlbcTest() {
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WebRtcIlbcfix_EncoderFree(encoder_);
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}
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virtual void InitEncoder() {
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ASSERT_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, 30)); // 30 ms.
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}
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virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
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uint8_t* output) {
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int enc_len_bytes =
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WebRtcIlbcfix_Encode(encoder_, input,
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static_cast<int>(input_len_samples),
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reinterpret_cast<int16_t*>(output));
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EXPECT_EQ(50, enc_len_bytes);
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return enc_len_bytes;
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}
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// Overload the default test since iLBC's function WebRtcIlbcfix_NetEqPlc does
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// not return any data. It simply resets a few states and returns 0.
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void DecodePlcTest() {
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InitEncoder();
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size_t enc_len = EncodeFrame(input_, frame_size_, encoded_);
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AudioDecoder::SpeechType speech_type;
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EXPECT_EQ(0, decoder_->Init());
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size_t dec_len =
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decoder_->Decode(encoded_, enc_len, decoded_, &speech_type);
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EXPECT_EQ(frame_size_, dec_len);
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// Simply call DecodePlc and verify that we get 0 as return value.
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EXPECT_EQ(0, decoder_->DecodePlc(1, decoded_));
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}
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iLBC_encinst_t* encoder_;
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};
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class AudioDecoderIsacFloatTest : public AudioDecoderTest {
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protected:
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AudioDecoderIsacFloatTest() : AudioDecoderTest() {
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input_size_ = 160;
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frame_size_ = 480;
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data_length_ = 10 * frame_size_;
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decoder_ = new AudioDecoderIsac;
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assert(decoder_);
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WebRtcIsac_Create(&encoder_);
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WebRtcIsac_SetEncSampRate(encoder_, 16000);
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}
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~AudioDecoderIsacFloatTest() {
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WebRtcIsac_Free(encoder_);
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}
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virtual void InitEncoder() {
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ASSERT_EQ(0, WebRtcIsac_EncoderInit(encoder_, 1)); // Fixed mode.
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ASSERT_EQ(0, WebRtcIsac_Control(encoder_, 32000, 30)); // 32 kbps, 30 ms.
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}
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virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
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uint8_t* output) {
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// Insert 3 * 10 ms. Expect non-zero output on third call.
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EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
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reinterpret_cast<int16_t*>(output)));
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input += input_size_;
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EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
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reinterpret_cast<int16_t*>(output)));
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input += input_size_;
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int enc_len_bytes =
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WebRtcIsac_Encode(encoder_, input, reinterpret_cast<int16_t*>(output));
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EXPECT_GT(enc_len_bytes, 0);
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return enc_len_bytes;
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}
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ISACStruct* encoder_;
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int input_size_;
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};
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class AudioDecoderIsacSwbTest : public AudioDecoderTest {
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protected:
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AudioDecoderIsacSwbTest() : AudioDecoderTest() {
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input_size_ = 320;
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frame_size_ = 960;
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data_length_ = 10 * frame_size_;
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decoder_ = new AudioDecoderIsacSwb;
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assert(decoder_);
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WebRtcIsac_Create(&encoder_);
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WebRtcIsac_SetEncSampRate(encoder_, 32000);
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}
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~AudioDecoderIsacSwbTest() {
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WebRtcIsac_Free(encoder_);
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}
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virtual void InitEncoder() {
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ASSERT_EQ(0, WebRtcIsac_EncoderInit(encoder_, 1)); // Fixed mode.
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ASSERT_EQ(0, WebRtcIsac_Control(encoder_, 32000, 30)); // 32 kbps, 30 ms.
|
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}
|
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|
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virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
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uint8_t* output) {
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// Insert 3 * 10 ms. Expect non-zero output on third call.
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EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
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reinterpret_cast<int16_t*>(output)));
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input += input_size_;
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EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
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reinterpret_cast<int16_t*>(output)));
|
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input += input_size_;
|
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int enc_len_bytes =
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WebRtcIsac_Encode(encoder_, input, reinterpret_cast<int16_t*>(output));
|
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EXPECT_GT(enc_len_bytes, 0);
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return enc_len_bytes;
|
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}
|
||||
|
||||
ISACStruct* encoder_;
|
||||
int input_size_;
|
||||
};
|
||||
|
||||
// This test is identical to AudioDecoderIsacSwbTest, except that it creates
|
||||
// an AudioDecoderIsacFb decoder object.
|
||||
class AudioDecoderIsacFbTest : public AudioDecoderIsacSwbTest {
|
||||
protected:
|
||||
AudioDecoderIsacFbTest() : AudioDecoderIsacSwbTest() {
|
||||
// Delete the |decoder_| that was created by AudioDecoderIsacSwbTest and
|
||||
// create an AudioDecoderIsacFb object instead.
|
||||
delete decoder_;
|
||||
decoder_ = new AudioDecoderIsacFb;
|
||||
assert(decoder_);
|
||||
}
|
||||
};
|
||||
|
||||
class AudioDecoderIsacFixTest : public AudioDecoderTest {
|
||||
protected:
|
||||
AudioDecoderIsacFixTest() : AudioDecoderTest() {
|
||||
input_size_ = 160;
|
||||
frame_size_ = 480;
|
||||
data_length_ = 10 * frame_size_;
|
||||
decoder_ = new AudioDecoderIsacFix;
|
||||
assert(decoder_);
|
||||
WebRtcIsacfix_Create(&encoder_);
|
||||
}
|
||||
|
||||
~AudioDecoderIsacFixTest() {
|
||||
WebRtcIsacfix_Free(encoder_);
|
||||
}
|
||||
|
||||
virtual void InitEncoder() {
|
||||
ASSERT_EQ(0, WebRtcIsacfix_EncoderInit(encoder_, 1)); // Fixed mode.
|
||||
ASSERT_EQ(0,
|
||||
WebRtcIsacfix_Control(encoder_, 32000, 30)); // 32 kbps, 30 ms.
|
||||
}
|
||||
|
||||
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
|
||||
uint8_t* output) {
|
||||
// Insert 3 * 10 ms. Expect non-zero output on third call.
|
||||
EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input,
|
||||
reinterpret_cast<int16_t*>(output)));
|
||||
input += input_size_;
|
||||
EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input,
|
||||
reinterpret_cast<int16_t*>(output)));
|
||||
input += input_size_;
|
||||
int enc_len_bytes = WebRtcIsacfix_Encode(
|
||||
encoder_, input, reinterpret_cast<int16_t*>(output));
|
||||
EXPECT_GT(enc_len_bytes, 0);
|
||||
return enc_len_bytes;
|
||||
}
|
||||
|
||||
ISACFIX_MainStruct* encoder_;
|
||||
int input_size_;
|
||||
};
|
||||
|
||||
class AudioDecoderG722Test : public AudioDecoderTest {
|
||||
protected:
|
||||
AudioDecoderG722Test() : AudioDecoderTest() {
|
||||
frame_size_ = 160;
|
||||
data_length_ = 10 * frame_size_;
|
||||
decoder_ = new AudioDecoderG722;
|
||||
assert(decoder_);
|
||||
WebRtcG722_CreateEncoder(&encoder_);
|
||||
}
|
||||
|
||||
~AudioDecoderG722Test() {
|
||||
WebRtcG722_FreeEncoder(encoder_);
|
||||
}
|
||||
|
||||
virtual void InitEncoder() {
|
||||
ASSERT_EQ(0, WebRtcG722_EncoderInit(encoder_));
|
||||
}
|
||||
|
||||
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
|
||||
uint8_t* output) {
|
||||
int enc_len_bytes =
|
||||
WebRtcG722_Encode(encoder_, const_cast<int16_t*>(input),
|
||||
static_cast<int>(input_len_samples),
|
||||
reinterpret_cast<int16_t*>(output));
|
||||
EXPECT_EQ(80, enc_len_bytes);
|
||||
return enc_len_bytes;
|
||||
}
|
||||
|
||||
G722EncInst* encoder_;
|
||||
};
|
||||
|
||||
class AudioDecoderG722StereoTest : public AudioDecoderG722Test {
|
||||
protected:
|
||||
AudioDecoderG722StereoTest() : AudioDecoderG722Test() {
|
||||
channels_ = 2;
|
||||
// Delete the |decoder_| that was created by AudioDecoderG722Test and
|
||||
// create an AudioDecoderG722Stereo object instead.
|
||||
delete decoder_;
|
||||
decoder_ = new AudioDecoderG722Stereo;
|
||||
assert(decoder_);
|
||||
}
|
||||
|
||||
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
|
||||
uint8_t* output) {
|
||||
uint8_t* temp_output = new uint8_t[data_length_ * 2];
|
||||
// Encode a mono payload using the base test class.
|
||||
int mono_enc_len_bytes =
|
||||
AudioDecoderG722Test::EncodeFrame(input, input_len_samples,
|
||||
temp_output);
|
||||
// The bit-stream consists of 4-bit samples:
|
||||
// +--------+--------+--------+
|
||||
// | s0 s1 | s2 s3 | s4 s5 |
|
||||
// +--------+--------+--------+
|
||||
//
|
||||
// Duplicate them to the |output| such that the stereo stream becomes:
|
||||
// +--------+--------+--------+
|
||||
// | s0 s0 | s1 s1 | s2 s2 |
|
||||
// +--------+--------+--------+
|
||||
EXPECT_LE(mono_enc_len_bytes * 2, static_cast<int>(data_length_ * 2));
|
||||
uint8_t* output_ptr = output;
|
||||
for (int i = 0; i < mono_enc_len_bytes; ++i) {
|
||||
*output_ptr = (temp_output[i] & 0xF0) + (temp_output[i] >> 4);
|
||||
++output_ptr;
|
||||
*output_ptr = (temp_output[i] << 4) + (temp_output[i] & 0x0F);
|
||||
++output_ptr;
|
||||
}
|
||||
delete [] temp_output;
|
||||
return mono_enc_len_bytes * 2;
|
||||
}
|
||||
};
|
||||
|
||||
#ifdef WEBRTC_CODEC_CELT
|
||||
class AudioDecoderCeltTest : public AudioDecoderTest {
|
||||
protected:
|
||||
static const int kEncodingRateBitsPerSecond = 64000;
|
||||
AudioDecoderCeltTest() : AudioDecoderTest(), encoder_(NULL) {
|
||||
frame_size_ = 640;
|
||||
data_length_ = 10 * frame_size_;
|
||||
decoder_ = AudioDecoder::CreateAudioDecoder(kDecoderCELT_32);
|
||||
assert(decoder_);
|
||||
WebRtcCelt_CreateEnc(&encoder_, static_cast<int>(channels_));
|
||||
}
|
||||
|
||||
~AudioDecoderCeltTest() {
|
||||
WebRtcCelt_FreeEnc(encoder_);
|
||||
}
|
||||
|
||||
virtual void InitEncoder() {
|
||||
assert(encoder_);
|
||||
ASSERT_EQ(0, WebRtcCelt_EncoderInit(
|
||||
encoder_, static_cast<int>(channels_), kEncodingRateBitsPerSecond));
|
||||
}
|
||||
|
||||
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
|
||||
uint8_t* output) {
|
||||
assert(encoder_);
|
||||
return WebRtcCelt_Encode(encoder_, input, output);
|
||||
}
|
||||
|
||||
CELT_encinst_t* encoder_;
|
||||
};
|
||||
|
||||
class AudioDecoderCeltStereoTest : public AudioDecoderTest {
|
||||
protected:
|
||||
static const int kEncodingRateBitsPerSecond = 64000;
|
||||
AudioDecoderCeltStereoTest() : AudioDecoderTest(), encoder_(NULL) {
|
||||
channels_ = 2;
|
||||
frame_size_ = 640;
|
||||
data_length_ = 10 * frame_size_;
|
||||
decoder_ = AudioDecoder::CreateAudioDecoder(kDecoderCELT_32_2ch);
|
||||
assert(decoder_);
|
||||
stereo_input_ = new int16_t[frame_size_ * channels_];
|
||||
WebRtcCelt_CreateEnc(&encoder_, static_cast<int>(channels_));
|
||||
}
|
||||
|
||||
~AudioDecoderCeltStereoTest() {
|
||||
delete [] stereo_input_;
|
||||
WebRtcCelt_FreeEnc(encoder_);
|
||||
}
|
||||
|
||||
virtual void InitEncoder() {
|
||||
assert(encoder_);
|
||||
ASSERT_EQ(0, WebRtcCelt_EncoderInit(
|
||||
encoder_, static_cast<int>(channels_), kEncodingRateBitsPerSecond));
|
||||
}
|
||||
|
||||
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
|
||||
uint8_t* output) {
|
||||
assert(encoder_);
|
||||
assert(stereo_input_);
|
||||
for (size_t n = 0; n < frame_size_; ++n) {
|
||||
stereo_input_[n * 2] = stereo_input_[n * 2 + 1] = input[n];
|
||||
}
|
||||
return WebRtcCelt_Encode(encoder_, stereo_input_, output);
|
||||
}
|
||||
|
||||
int16_t* stereo_input_;
|
||||
CELT_encinst_t* encoder_;
|
||||
};
|
||||
|
||||
#endif
|
||||
|
||||
class AudioDecoderOpusTest : public AudioDecoderTest {
|
||||
protected:
|
||||
AudioDecoderOpusTest() : AudioDecoderTest() {
|
||||
frame_size_ = 320;
|
||||
data_length_ = 10 * frame_size_;
|
||||
decoder_ = new AudioDecoderOpus(kDecoderOpus);
|
||||
assert(decoder_);
|
||||
WebRtcOpus_EncoderCreate(&encoder_, 1);
|
||||
}
|
||||
|
||||
~AudioDecoderOpusTest() {
|
||||
WebRtcOpus_EncoderFree(encoder_);
|
||||
}
|
||||
|
||||
virtual void InitEncoder() {}
|
||||
|
||||
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
|
||||
uint8_t* output) {
|
||||
// Upsample from 32 to 48 kHz.
|
||||
Resampler rs;
|
||||
rs.Reset(32000, 48000, kResamplerSynchronous);
|
||||
const int max_resamp_len_samples = static_cast<int>(input_len_samples) *
|
||||
3 / 2;
|
||||
int16_t* resamp_input = new int16_t[max_resamp_len_samples];
|
||||
int resamp_len_samples;
|
||||
EXPECT_EQ(0, rs.Push(input, static_cast<int>(input_len_samples),
|
||||
resamp_input, max_resamp_len_samples,
|
||||
resamp_len_samples));
|
||||
EXPECT_EQ(max_resamp_len_samples, resamp_len_samples);
|
||||
int enc_len_bytes =
|
||||
WebRtcOpus_Encode(encoder_, resamp_input, resamp_len_samples,
|
||||
static_cast<int>(data_length_), output);
|
||||
EXPECT_GT(enc_len_bytes, 0);
|
||||
delete [] resamp_input;
|
||||
return enc_len_bytes;
|
||||
}
|
||||
|
||||
OpusEncInst* encoder_;
|
||||
};
|
||||
|
||||
class AudioDecoderOpusStereoTest : public AudioDecoderTest {
|
||||
protected:
|
||||
AudioDecoderOpusStereoTest() : AudioDecoderTest() {
|
||||
channels_ = 2;
|
||||
frame_size_ = 320;
|
||||
data_length_ = 10 * frame_size_;
|
||||
decoder_ = new AudioDecoderOpus(kDecoderOpus_2ch);
|
||||
assert(decoder_);
|
||||
WebRtcOpus_EncoderCreate(&encoder_, 2);
|
||||
}
|
||||
|
||||
~AudioDecoderOpusStereoTest() {
|
||||
WebRtcOpus_EncoderFree(encoder_);
|
||||
}
|
||||
|
||||
virtual void InitEncoder() {}
|
||||
|
||||
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
|
||||
uint8_t* output) {
|
||||
// Create stereo by duplicating each sample in |input|.
|
||||
const int input_stereo_samples = static_cast<int>(input_len_samples) * 2;
|
||||
int16_t* input_stereo = new int16_t[input_stereo_samples];
|
||||
for (size_t i = 0; i < input_len_samples; i++)
|
||||
input_stereo[i * 2] = input_stereo[i * 2 + 1] = input[i];
|
||||
// Upsample from 32 to 48 kHz.
|
||||
Resampler rs;
|
||||
rs.Reset(32000, 48000, kResamplerSynchronousStereo);
|
||||
const int max_resamp_len_samples = input_stereo_samples * 3 / 2;
|
||||
int16_t* resamp_input = new int16_t[max_resamp_len_samples];
|
||||
int resamp_len_samples;
|
||||
EXPECT_EQ(0, rs.Push(input_stereo, input_stereo_samples, resamp_input,
|
||||
max_resamp_len_samples, resamp_len_samples));
|
||||
EXPECT_EQ(max_resamp_len_samples, resamp_len_samples);
|
||||
int enc_len_bytes =
|
||||
WebRtcOpus_Encode(encoder_, resamp_input, resamp_len_samples / 2,
|
||||
static_cast<int16_t>(data_length_), output);
|
||||
EXPECT_GT(enc_len_bytes, 0);
|
||||
delete [] resamp_input;
|
||||
delete [] input_stereo;
|
||||
return enc_len_bytes;
|
||||
}
|
||||
|
||||
OpusEncInst* encoder_;
|
||||
};
|
||||
|
||||
TEST_F(AudioDecoderPcmUTest, EncodeDecode) {
|
||||
int tolerance = 251;
|
||||
double mse = 1734.0;
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCMu));
|
||||
EncodeDecodeTest(data_length_, tolerance, mse);
|
||||
ReInitTest();
|
||||
EXPECT_FALSE(decoder_->HasDecodePlc());
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderPcmATest, EncodeDecode) {
|
||||
int tolerance = 308;
|
||||
double mse = 1931.0;
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCMa));
|
||||
EncodeDecodeTest(data_length_, tolerance, mse);
|
||||
ReInitTest();
|
||||
EXPECT_FALSE(decoder_->HasDecodePlc());
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderPcm16BTest, EncodeDecode) {
|
||||
int tolerance = 0;
|
||||
double mse = 0.0;
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16B));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16Bwb));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16Bswb32kHz));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16Bswb48kHz));
|
||||
EncodeDecodeTest(2 * data_length_, tolerance, mse);
|
||||
ReInitTest();
|
||||
EXPECT_FALSE(decoder_->HasDecodePlc());
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderIlbcTest, EncodeDecode) {
|
||||
int tolerance = 6808;
|
||||
double mse = 2.13e6;
|
||||
int delay = 80; // Delay from input to output.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderILBC));
|
||||
EncodeDecodeTest(500, tolerance, mse, delay);
|
||||
ReInitTest();
|
||||
EXPECT_TRUE(decoder_->HasDecodePlc());
|
||||
DecodePlcTest();
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderIsacFloatTest, EncodeDecode) {
|
||||
int tolerance = 3399;
|
||||
double mse = 434951.0;
|
||||
int delay = 48; // Delay from input to output.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderISAC));
|
||||
EncodeDecodeTest(0, tolerance, mse, delay);
|
||||
ReInitTest();
|
||||
EXPECT_TRUE(decoder_->HasDecodePlc());
|
||||
DecodePlcTest();
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderIsacSwbTest, EncodeDecode) {
|
||||
int tolerance = 19757;
|
||||
double mse = 8.18e6;
|
||||
int delay = 160; // Delay from input to output.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderISACswb));
|
||||
EncodeDecodeTest(0, tolerance, mse, delay);
|
||||
ReInitTest();
|
||||
EXPECT_TRUE(decoder_->HasDecodePlc());
|
||||
DecodePlcTest();
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderIsacFbTest, EncodeDecode) {
|
||||
int tolerance = 19757;
|
||||
double mse = 8.18e6;
|
||||
int delay = 160; // Delay from input to output.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderISACswb));
|
||||
EncodeDecodeTest(0, tolerance, mse, delay);
|
||||
ReInitTest();
|
||||
EXPECT_TRUE(decoder_->HasDecodePlc());
|
||||
DecodePlcTest();
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderIsacFixTest, DISABLED_EncodeDecode) {
|
||||
int tolerance = 11034;
|
||||
double mse = 3.46e6;
|
||||
int delay = 54; // Delay from input to output.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderISAC));
|
||||
EncodeDecodeTest(735, tolerance, mse, delay);
|
||||
ReInitTest();
|
||||
EXPECT_FALSE(decoder_->HasDecodePlc());
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderG722Test, EncodeDecode) {
|
||||
int tolerance = 6176;
|
||||
double mse = 238630.0;
|
||||
int delay = 22; // Delay from input to output.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderG722));
|
||||
EncodeDecodeTest(data_length_ / 2, tolerance, mse, delay);
|
||||
ReInitTest();
|
||||
EXPECT_FALSE(decoder_->HasDecodePlc());
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderG722StereoTest, CreateAndDestroy) {
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderG722_2ch));
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderG722StereoTest, EncodeDecode) {
|
||||
int tolerance = 6176;
|
||||
int channel_diff_tolerance = 0;
|
||||
double mse = 238630.0;
|
||||
int delay = 22; // Delay from input to output.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderG722_2ch));
|
||||
EncodeDecodeTest(data_length_, tolerance, mse, delay, channel_diff_tolerance);
|
||||
ReInitTest();
|
||||
EXPECT_FALSE(decoder_->HasDecodePlc());
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderOpusTest, EncodeDecode) {
|
||||
int tolerance = 6176;
|
||||
double mse = 238630.0;
|
||||
int delay = 22; // Delay from input to output.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderOpus));
|
||||
EncodeDecodeTest(0, tolerance, mse, delay);
|
||||
ReInitTest();
|
||||
EXPECT_FALSE(decoder_->HasDecodePlc());
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderOpusStereoTest, EncodeDecode) {
|
||||
int tolerance = 6176;
|
||||
int channel_diff_tolerance = 0;
|
||||
double mse = 238630.0;
|
||||
int delay = 22; // Delay from input to output.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderOpus_2ch));
|
||||
EncodeDecodeTest(0, tolerance, mse, delay, channel_diff_tolerance);
|
||||
ReInitTest();
|
||||
EXPECT_FALSE(decoder_->HasDecodePlc());
|
||||
}
|
||||
|
||||
#ifdef WEBRTC_CODEC_CELT
|
||||
// In the two following CELT tests, the low amplitude of the test signal allow
|
||||
// us to have such low error thresholds, i.e. |tolerance|, |mse|. Furthermore,
|
||||
// in general, stereo signals with identical channels do not result in identical
|
||||
// encoded channels.
|
||||
TEST_F(AudioDecoderCeltTest, EncodeDecode) {
|
||||
int tolerance = 20;
|
||||
double mse = 17.0;
|
||||
int delay = 80; // Delay from input to output in samples.
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCELT_32));
|
||||
EncodeDecodeTest(1600, tolerance, mse, delay);
|
||||
ReInitTest();
|
||||
EXPECT_TRUE(decoder_->HasDecodePlc());
|
||||
DecodePlcTest();
|
||||
}
|
||||
|
||||
TEST_F(AudioDecoderCeltStereoTest, EncodeDecode) {
|
||||
int tolerance = 20;
|
||||
// If both channels are identical, CELT not necessarily decodes identical
|
||||
// channels. However, for this input this is the case.
|
||||
int channel_diff_tolerance = 0;
|
||||
double mse = 20.0;
|
||||
// Delay from input to output in samples, accounting for stereo.
|
||||
int delay = 160;
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCELT_32_2ch));
|
||||
EncodeDecodeTest(1600, tolerance, mse, delay, channel_diff_tolerance);
|
||||
ReInitTest();
|
||||
EXPECT_TRUE(decoder_->HasDecodePlc());
|
||||
DecodePlcTest();
|
||||
}
|
||||
#endif
|
||||
|
||||
TEST(AudioDecoder, CodecSampleRateHz) {
|
||||
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderPCMu));
|
||||
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderPCMa));
|
||||
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderPCMu_2ch));
|
||||
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderPCMa_2ch));
|
||||
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderILBC));
|
||||
EXPECT_EQ(16000, AudioDecoder::CodecSampleRateHz(kDecoderISAC));
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderISACswb));
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderISACfb));
|
||||
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderPCM16B));
|
||||
EXPECT_EQ(16000, AudioDecoder::CodecSampleRateHz(kDecoderPCM16Bwb));
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderPCM16Bswb32kHz));
|
||||
EXPECT_EQ(48000, AudioDecoder::CodecSampleRateHz(kDecoderPCM16Bswb48kHz));
|
||||
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderPCM16B_2ch));
|
||||
EXPECT_EQ(16000, AudioDecoder::CodecSampleRateHz(kDecoderPCM16Bwb_2ch));
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderPCM16Bswb32kHz_2ch));
|
||||
EXPECT_EQ(48000, AudioDecoder::CodecSampleRateHz(kDecoderPCM16Bswb48kHz_2ch));
|
||||
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderPCM16B_5ch));
|
||||
EXPECT_EQ(16000, AudioDecoder::CodecSampleRateHz(kDecoderG722));
|
||||
EXPECT_EQ(16000, AudioDecoder::CodecSampleRateHz(kDecoderG722_2ch));
|
||||
EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderRED));
|
||||
EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderAVT));
|
||||
EXPECT_EQ(8000, AudioDecoder::CodecSampleRateHz(kDecoderCNGnb));
|
||||
EXPECT_EQ(16000, AudioDecoder::CodecSampleRateHz(kDecoderCNGwb));
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCNGswb32kHz));
|
||||
// TODO(tlegrand): Change 32000 to 48000 below once ACM has 48 kHz support.
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCNGswb48kHz));
|
||||
EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderArbitrary));
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderOpus));
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderOpus_2ch));
|
||||
#ifdef WEBRTC_CODEC_CELT
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32));
|
||||
EXPECT_EQ(32000, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32_2ch));
|
||||
#else
|
||||
EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32));
|
||||
EXPECT_EQ(-1, AudioDecoder::CodecSampleRateHz(kDecoderCELT_32_2ch));
|
||||
#endif
|
||||
}
|
||||
|
||||
TEST(AudioDecoder, CodecSupported) {
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCMu));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCMa));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCMu_2ch));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCMa_2ch));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderILBC));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderISAC));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderISACswb));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderISACfb));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16B));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16Bwb));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16Bswb32kHz));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16Bswb48kHz));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16B_2ch));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16Bwb_2ch));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16Bswb32kHz_2ch));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16Bswb48kHz_2ch));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderPCM16B_5ch));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderG722));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderG722_2ch));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderRED));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderAVT));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCNGnb));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCNGwb));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCNGswb32kHz));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCNGswb48kHz));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderArbitrary));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderOpus));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderOpus_2ch));
|
||||
#ifdef WEBRTC_CODEC_CELT
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCELT_32));
|
||||
EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderCELT_32_2ch));
|
||||
#else
|
||||
EXPECT_FALSE(AudioDecoder::CodecSupported(kDecoderCELT_32));
|
||||
EXPECT_FALSE(AudioDecoder::CodecSupported(kDecoderCELT_32_2ch));
|
||||
#endif
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
Reference in New Issue
Block a user