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https://github.com/oxen-io/session-android.git
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388 lines
12 KiB
C++
388 lines
12 KiB
C++
/*
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* Copyright (c) 2013 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/main/test/opus_test.h"
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#include <assert.h>
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#include <string>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/common_types.h"
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#include "webrtc/engine_configurations.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/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_opus.h"
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#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
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#include "webrtc/modules/audio_coding/main/test/utility.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#include "webrtc/test/testsupport/fileutils.h"
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namespace webrtc {
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OpusTest::OpusTest()
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: acm_receiver_(AudioCodingModule::Create(0)),
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channel_a2b_(NULL),
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counter_(0),
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payload_type_(255),
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rtp_timestamp_(0) {}
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OpusTest::~OpusTest() {
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if (channel_a2b_ != NULL) {
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delete channel_a2b_;
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channel_a2b_ = NULL;
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}
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if (opus_mono_encoder_ != NULL) {
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WebRtcOpus_EncoderFree(opus_mono_encoder_);
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opus_mono_encoder_ = NULL;
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}
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if (opus_stereo_encoder_ != NULL) {
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WebRtcOpus_EncoderFree(opus_stereo_encoder_);
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opus_stereo_encoder_ = NULL;
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}
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if (opus_mono_decoder_ != NULL) {
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WebRtcOpus_DecoderFree(opus_mono_decoder_);
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opus_mono_decoder_ = NULL;
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}
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if (opus_stereo_decoder_ != NULL) {
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WebRtcOpus_DecoderFree(opus_stereo_decoder_);
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opus_stereo_decoder_ = NULL;
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}
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}
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void OpusTest::Perform() {
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#ifndef WEBRTC_CODEC_OPUS
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// Opus isn't defined, exit.
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return;
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#else
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uint16_t frequency_hz;
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int audio_channels;
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int16_t test_cntr = 0;
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// Open both mono and stereo test files in 32 kHz.
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const std::string file_name_stereo =
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webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
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const std::string file_name_mono =
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webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
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frequency_hz = 32000;
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in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb");
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in_file_stereo_.ReadStereo(true);
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in_file_mono_.Open(file_name_mono, frequency_hz, "rb");
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in_file_mono_.ReadStereo(false);
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// Create Opus encoders for mono and stereo.
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ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1), -1);
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ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2), -1);
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// Create Opus decoders for mono and stereo for stand-alone testing of Opus.
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ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1);
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ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1);
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ASSERT_GT(WebRtcOpus_DecoderInitNew(opus_mono_decoder_), -1);
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ASSERT_GT(WebRtcOpus_DecoderInitNew(opus_stereo_decoder_), -1);
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ASSERT_TRUE(acm_receiver_.get() != NULL);
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EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
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// Register Opus stereo as receiving codec.
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CodecInst opus_codec_param;
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int codec_id = acm_receiver_->Codec("opus", 48000, 2);
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EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
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payload_type_ = opus_codec_param.pltype;
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EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
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// Create and connect the channel.
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channel_a2b_ = new TestPackStereo;
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channel_a2b_->RegisterReceiverACM(acm_receiver_.get());
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//
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// Test Stereo.
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//
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channel_a2b_->set_codec_mode(kStereo);
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audio_channels = 2;
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test_cntr++;
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OpenOutFile(test_cntr);
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// Run Opus with 2.5 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 120);
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// Run Opus with 5 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 240);
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// Run Opus with 10 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 480);
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// Run Opus with 20 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 960);
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// Run Opus with 40 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 1920);
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// Run Opus with 60 ms frame size.
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Run(channel_a2b_, audio_channels, 64000, 2880);
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out_file_.Close();
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out_file_standalone_.Close();
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//
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// Test Opus stereo with packet-losses.
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//
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test_cntr++;
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OpenOutFile(test_cntr);
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// Run Opus with 20 ms frame size, 1% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 1);
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// Run Opus with 20 ms frame size, 5% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 5);
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// Run Opus with 20 ms frame size, 10% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 10);
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out_file_.Close();
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out_file_standalone_.Close();
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//
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// Test Mono.
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//
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channel_a2b_->set_codec_mode(kMono);
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audio_channels = 1;
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test_cntr++;
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OpenOutFile(test_cntr);
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// Register Opus mono as receiving codec.
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opus_codec_param.channels = 1;
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EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
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// Run Opus with 2.5 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 120);
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// Run Opus with 5 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 240);
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// Run Opus with 10 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 480);
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// Run Opus with 20 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 960);
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// Run Opus with 40 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 1920);
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// Run Opus with 60 ms frame size.
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Run(channel_a2b_, audio_channels, 32000, 2880);
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out_file_.Close();
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out_file_standalone_.Close();
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//
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// Test Opus mono with packet-losses.
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//
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test_cntr++;
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OpenOutFile(test_cntr);
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// Run Opus with 20 ms frame size, 1% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 1);
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// Run Opus with 20 ms frame size, 5% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 5);
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// Run Opus with 20 ms frame size, 10% packet loss.
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Run(channel_a2b_, audio_channels, 64000, 960, 10);
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// Close the files.
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in_file_stereo_.Close();
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in_file_mono_.Close();
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out_file_.Close();
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out_file_standalone_.Close();
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#endif
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}
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void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
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int frame_length, int percent_loss) {
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AudioFrame audio_frame;
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int32_t out_freq_hz_b = out_file_.SamplingFrequency();
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const int kBufferSizeSamples = 480 * 12 * 2; // Can hold 120 ms stereo audio.
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int16_t audio[kBufferSizeSamples];
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int16_t out_audio[kBufferSizeSamples];
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int16_t audio_type;
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int written_samples = 0;
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int read_samples = 0;
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int decoded_samples = 0;
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bool first_packet = true;
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uint32_t start_time_stamp = 0;
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channel->reset_payload_size();
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counter_ = 0;
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// Set encoder rate.
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EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
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EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));
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#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
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// If we are on Android, iOS and/or ARM, use a lower complexity setting as
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// default.
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const int kOpusComplexity5 = 5;
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EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
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EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_,
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kOpusComplexity5));
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#endif
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// Make sure the runtime is less than 60 seconds to pass Android test.
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for (size_t audio_length = 0; audio_length < 10000; audio_length += 10) {
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bool lost_packet = false;
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// Get 10 msec of audio.
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if (channels == 1) {
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if (in_file_mono_.EndOfFile()) {
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break;
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}
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in_file_mono_.Read10MsData(audio_frame);
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} else {
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if (in_file_stereo_.EndOfFile()) {
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break;
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}
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in_file_stereo_.Read10MsData(audio_frame);
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}
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// If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
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EXPECT_EQ(480,
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resampler_.Resample10Msec(audio_frame.data_,
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audio_frame.sample_rate_hz_,
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48000,
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channels,
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kBufferSizeSamples - written_samples,
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&audio[written_samples]));
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written_samples += 480 * channels;
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// Sometimes we need to loop over the audio vector to produce the right
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// number of packets.
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int loop_encode = (written_samples - read_samples) /
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(channels * frame_length);
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if (loop_encode > 0) {
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const int kMaxBytes = 1000; // Maximum number of bytes for one packet.
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int16_t bitstream_len_byte;
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uint8_t bitstream[kMaxBytes];
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for (int i = 0; i < loop_encode; i++) {
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if (channels == 1) {
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bitstream_len_byte = WebRtcOpus_Encode(
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opus_mono_encoder_, &audio[read_samples],
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frame_length, kMaxBytes, bitstream);
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ASSERT_GT(bitstream_len_byte, -1);
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} else {
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bitstream_len_byte = WebRtcOpus_Encode(
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opus_stereo_encoder_, &audio[read_samples],
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frame_length, kMaxBytes, bitstream);
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ASSERT_GT(bitstream_len_byte, -1);
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}
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// Simulate packet loss by setting |packet_loss_| to "true" in
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// |percent_loss| percent of the loops.
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// TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
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if (percent_loss > 0) {
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if (counter_ == floor((100 / percent_loss) + 0.5)) {
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counter_ = 0;
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lost_packet = true;
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channel->set_lost_packet(true);
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} else {
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lost_packet = false;
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channel->set_lost_packet(false);
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}
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counter_++;
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}
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// Run stand-alone Opus decoder, or decode PLC.
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if (channels == 1) {
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if (!lost_packet) {
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decoded_samples += WebRtcOpus_DecodeNew(
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opus_mono_decoder_, bitstream, bitstream_len_byte,
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&out_audio[decoded_samples * channels], &audio_type);
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} else {
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decoded_samples += WebRtcOpus_DecodePlc(
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opus_mono_decoder_, &out_audio[decoded_samples * channels], 1);
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}
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} else {
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if (!lost_packet) {
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decoded_samples += WebRtcOpus_DecodeNew(
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opus_stereo_decoder_, bitstream, bitstream_len_byte,
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&out_audio[decoded_samples * channels], &audio_type);
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} else {
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decoded_samples += WebRtcOpus_DecodePlc(
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opus_stereo_decoder_, &out_audio[decoded_samples * channels],
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1);
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}
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}
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// Send data to the channel. "channel" will handle the loss simulation.
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channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_,
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bitstream, bitstream_len_byte, NULL);
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if (first_packet) {
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first_packet = false;
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start_time_stamp = rtp_timestamp_;
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}
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rtp_timestamp_ += frame_length;
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read_samples += frame_length * channels;
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}
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if (read_samples == written_samples) {
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read_samples = 0;
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written_samples = 0;
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}
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}
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// Run received side of ACM.
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ASSERT_EQ(0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
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// Write output speech to file.
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out_file_.Write10MsData(
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audio_frame.data_,
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audio_frame.samples_per_channel_ * audio_frame.num_channels_);
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// Write stand-alone speech to file.
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out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
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if (audio_frame.timestamp_ > start_time_stamp) {
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// Number of channels should be the same for both stand-alone and
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// ACM-decoding.
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EXPECT_EQ(audio_frame.num_channels_, channels);
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}
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decoded_samples = 0;
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}
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if (in_file_mono_.EndOfFile()) {
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in_file_mono_.Rewind();
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}
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if (in_file_stereo_.EndOfFile()) {
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in_file_stereo_.Rewind();
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}
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// Reset in case we ended with a lost packet.
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channel->set_lost_packet(false);
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}
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void OpusTest::OpenOutFile(int test_number) {
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std::string file_name;
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std::stringstream file_stream;
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file_stream << webrtc::test::OutputPath() << "opustest_out_"
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<< test_number << ".pcm";
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file_name = file_stream.str();
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out_file_.Open(file_name, 48000, "wb");
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file_stream.str("");
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file_name = file_stream.str();
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file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
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<< test_number << ".pcm";
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file_name = file_stream.str();
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out_file_standalone_.Open(file_name, 48000, "wb");
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}
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} // namespace webrtc
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