600 lines
21 KiB
C++
600 lines
21 KiB
C++
/*
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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 "modules/audio_coding/test/TestStereo.h"
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#include <string>
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#include "absl/strings/match.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
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#include "modules/include/module_common_types.h"
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#include "rtc_base/strings/string_builder.h"
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#include "test/gtest.h"
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#include "test/testsupport/file_utils.h"
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namespace webrtc {
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// Class for simulating packet handling
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TestPackStereo::TestPackStereo()
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: receiver_acm_(NULL),
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seq_no_(0),
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timestamp_diff_(0),
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last_in_timestamp_(0),
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total_bytes_(0),
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payload_size_(0),
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lost_packet_(false) {}
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TestPackStereo::~TestPackStereo() {}
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void TestPackStereo::RegisterReceiverACM(AudioCodingModule* acm) {
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receiver_acm_ = acm;
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return;
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}
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int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
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const uint8_t payload_type,
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const uint32_t timestamp,
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const uint8_t* payload_data,
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const size_t payload_size,
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int64_t absolute_capture_timestamp_ms) {
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RTPHeader rtp_header;
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int32_t status = 0;
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rtp_header.markerBit = false;
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rtp_header.ssrc = 0;
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rtp_header.sequenceNumber = seq_no_++;
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rtp_header.payloadType = payload_type;
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rtp_header.timestamp = timestamp;
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if (frame_type == AudioFrameType::kEmptyFrame) {
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// Skip this frame
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return 0;
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}
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if (lost_packet_ == false) {
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status =
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receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_header);
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if (frame_type != AudioFrameType::kAudioFrameCN) {
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payload_size_ = static_cast<int>(payload_size);
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} else {
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payload_size_ = -1;
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}
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timestamp_diff_ = timestamp - last_in_timestamp_;
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last_in_timestamp_ = timestamp;
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total_bytes_ += payload_size;
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}
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return status;
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}
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uint16_t TestPackStereo::payload_size() {
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return static_cast<uint16_t>(payload_size_);
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}
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uint32_t TestPackStereo::timestamp_diff() {
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return timestamp_diff_;
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}
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void TestPackStereo::reset_payload_size() {
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payload_size_ = 0;
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}
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void TestPackStereo::set_codec_mode(enum StereoMonoMode mode) {
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codec_mode_ = mode;
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}
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void TestPackStereo::set_lost_packet(bool lost) {
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lost_packet_ = lost;
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}
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TestStereo::TestStereo()
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: acm_a_(AudioCodingModule::Create(
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AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
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acm_b_(AudioCodingModule::Create(
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AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
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channel_a2b_(NULL),
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test_cntr_(0),
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pack_size_samp_(0),
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pack_size_bytes_(0),
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counter_(0) {}
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TestStereo::~TestStereo() {
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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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}
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void TestStereo::Perform() {
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uint16_t frequency_hz;
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int audio_channels;
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int codec_channels;
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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_ = new PCMFile();
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in_file_mono_ = new PCMFile();
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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 and initialize two ACMs, one for each side of a one-to-one call.
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ASSERT_TRUE((acm_a_.get() != NULL) && (acm_b_.get() != NULL));
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EXPECT_EQ(0, acm_a_->InitializeReceiver());
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EXPECT_EQ(0, acm_b_->InitializeReceiver());
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acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
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{104, {"ISAC", 32000, 1}},
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{107, {"L16", 8000, 1}},
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{108, {"L16", 16000, 1}},
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{109, {"L16", 32000, 1}},
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{111, {"L16", 8000, 2}},
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{112, {"L16", 16000, 2}},
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{113, {"L16", 32000, 2}},
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{0, {"PCMU", 8000, 1}},
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{110, {"PCMU", 8000, 2}},
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{8, {"PCMA", 8000, 1}},
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{118, {"PCMA", 8000, 2}},
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{102, {"ILBC", 8000, 1}},
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{9, {"G722", 8000, 1}},
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{119, {"G722", 8000, 2}},
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{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
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{13, {"CN", 8000, 1}},
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{98, {"CN", 16000, 1}},
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{99, {"CN", 32000, 1}}});
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// Create and connect the channel.
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channel_a2b_ = new TestPackStereo;
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EXPECT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_));
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channel_a2b_->RegisterReceiverACM(acm_b_.get());
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char codec_pcma_temp[] = "PCMA";
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RegisterSendCodec('A', codec_pcma_temp, 8000, 64000, 80, 2);
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//
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// Test Stereo-To-Stereo for all codecs.
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//
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audio_channels = 2;
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codec_channels = 2;
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// All codecs are tested for all allowed sampling frequencies, rates and
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// packet sizes.
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channel_a2b_->set_codec_mode(kStereo);
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test_cntr_++;
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OpenOutFile(test_cntr_);
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char codec_g722[] = "G722";
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RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_g722, 16000, 64000, 320, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_g722, 16000, 64000, 480, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_g722, 16000, 64000, 640, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_g722, 16000, 64000, 800, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_g722, 16000, 64000, 960, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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channel_a2b_->set_codec_mode(kStereo);
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test_cntr_++;
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OpenOutFile(test_cntr_);
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char codec_l16[] = "L16";
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RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_l16, 8000, 128000, 160, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_l16, 8000, 128000, 240, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_l16, 8000, 128000, 320, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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test_cntr_++;
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_l16, 16000, 256000, 320, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_l16, 16000, 256000, 480, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_l16, 16000, 256000, 640, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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test_cntr_++;
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_l16, 32000, 512000, 640, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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#ifdef PCMA_AND_PCMU
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channel_a2b_->set_codec_mode(kStereo);
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audio_channels = 2;
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codec_channels = 2;
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test_cntr_++;
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OpenOutFile(test_cntr_);
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char codec_pcma[] = "PCMA";
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RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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test_cntr_++;
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OpenOutFile(test_cntr_);
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char codec_pcmu[] = "PCMU";
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RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcmu, 8000, 64000, 480, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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channel_a2b_->set_codec_mode(kStereo);
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audio_channels = 2;
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codec_channels = 2;
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test_cntr_++;
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OpenOutFile(test_cntr_);
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char codec_opus[] = "opus";
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// Run Opus with 10 ms frame size.
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RegisterSendCodec('A', codec_opus, 48000, 64000, 480, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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// Run Opus with 20 ms frame size.
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RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 2, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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// Run Opus with 40 ms frame size.
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RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 4, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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// Run Opus with 60 ms frame size.
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RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 6, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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// Run Opus with 20 ms frame size and different bitrates.
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RegisterSendCodec('A', codec_opus, 48000, 40000, 960, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_opus, 48000, 510000, 960, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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#endif
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//
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// Test Mono-To-Stereo for all codecs.
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//
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audio_channels = 1;
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codec_channels = 2;
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test_cntr_++;
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channel_a2b_->set_codec_mode(kStereo);
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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test_cntr_++;
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channel_a2b_->set_codec_mode(kStereo);
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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test_cntr_++;
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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test_cntr_++;
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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#ifdef PCMA_AND_PCMU
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test_cntr_++;
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channel_a2b_->set_codec_mode(kStereo);
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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// Keep encode and decode in stereo.
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test_cntr_++;
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channel_a2b_->set_codec_mode(kStereo);
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_opus, 48000, 64000, 960, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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// Encode in mono, decode in stereo mode.
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RegisterSendCodec('A', codec_opus, 48000, 64000, 960, 1);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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#endif
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//
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// Test Stereo-To-Mono for all codecs.
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//
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audio_channels = 2;
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codec_channels = 1;
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channel_a2b_->set_codec_mode(kMono);
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// Run stereo audio and mono codec.
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test_cntr_++;
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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test_cntr_++;
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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test_cntr_++;
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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test_cntr_++;
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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#ifdef PCMA_AND_PCMU
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test_cntr_++;
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OpenOutFile(test_cntr_);
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RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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test_cntr_++;
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OpenOutFile(test_cntr_);
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// Encode and decode in mono.
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RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels);
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acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
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Run(channel_a2b_, audio_channels, codec_channels);
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// Encode in stereo, decode in mono.
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RegisterSendCodec('A', codec_opus, 48000, 32000, 960, 2);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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// Test switching between decoding mono and stereo for Opus.
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// Decode in mono.
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test_cntr_++;
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OpenOutFile(test_cntr_);
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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// Decode in stereo.
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test_cntr_++;
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OpenOutFile(test_cntr_);
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acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
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Run(channel_a2b_, audio_channels, 2);
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out_file_.Close();
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// Decode in mono.
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test_cntr_++;
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OpenOutFile(test_cntr_);
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acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
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Run(channel_a2b_, audio_channels, codec_channels);
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out_file_.Close();
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#endif
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// Delete the file pointers.
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delete in_file_stereo_;
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delete in_file_mono_;
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}
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// Register Codec to use in the test
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//
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// Input: side - which ACM to use, 'A' or 'B'
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// codec_name - name to use when register the codec
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// sampling_freq_hz - sampling frequency in Herz
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// rate - bitrate in bytes
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// pack_size - packet size in samples
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// channels - number of channels; 1 for mono, 2 for stereo
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void TestStereo::RegisterSendCodec(char side,
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char* codec_name,
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int32_t sampling_freq_hz,
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int rate,
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int pack_size,
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int channels) {
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|
// Store packet size in samples, used to validate the received packet
|
|
pack_size_samp_ = pack_size;
|
|
|
|
// Store the expected packet size in bytes, used to validate the received
|
|
// packet. Add 0.875 to always round up to a whole byte.
|
|
pack_size_bytes_ = (uint16_t)(static_cast<float>(pack_size * rate) /
|
|
static_cast<float>(sampling_freq_hz * 8) +
|
|
0.875);
|
|
|
|
// Set pointer to the ACM where to register the codec
|
|
AudioCodingModule* my_acm = NULL;
|
|
switch (side) {
|
|
case 'A': {
|
|
my_acm = acm_a_.get();
|
|
break;
|
|
}
|
|
case 'B': {
|
|
my_acm = acm_b_.get();
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
ASSERT_TRUE(my_acm != NULL);
|
|
|
|
auto encoder_factory = CreateBuiltinAudioEncoderFactory();
|
|
const int clockrate_hz = absl::EqualsIgnoreCase(codec_name, "g722")
|
|
? sampling_freq_hz / 2
|
|
: sampling_freq_hz;
|
|
const std::string ptime = rtc::ToString(rtc::CheckedDivExact(
|
|
pack_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
|
|
SdpAudioFormat::Parameters params = {{"ptime", ptime}};
|
|
RTC_CHECK(channels == 1 || channels == 2);
|
|
if (absl::EqualsIgnoreCase(codec_name, "opus")) {
|
|
if (channels == 2) {
|
|
params["stereo"] = "1";
|
|
}
|
|
channels = 2;
|
|
params["maxaveragebitrate"] = rtc::ToString(rate);
|
|
}
|
|
constexpr int payload_type = 17;
|
|
auto encoder = encoder_factory->MakeAudioEncoder(
|
|
payload_type, SdpAudioFormat(codec_name, clockrate_hz, channels, params),
|
|
absl::nullopt);
|
|
EXPECT_NE(nullptr, encoder);
|
|
my_acm->SetEncoder(std::move(encoder));
|
|
|
|
send_codec_name_ = codec_name;
|
|
}
|
|
|
|
void TestStereo::Run(TestPackStereo* channel,
|
|
int in_channels,
|
|
int out_channels,
|
|
int percent_loss) {
|
|
AudioFrame audio_frame;
|
|
|
|
int32_t out_freq_hz_b = out_file_.SamplingFrequency();
|
|
uint16_t rec_size;
|
|
uint32_t time_stamp_diff;
|
|
channel->reset_payload_size();
|
|
int error_count = 0;
|
|
int variable_bytes = 0;
|
|
int variable_packets = 0;
|
|
// Set test length to 500 ms (50 blocks of 10 ms each).
|
|
in_file_mono_->SetNum10MsBlocksToRead(50);
|
|
in_file_stereo_->SetNum10MsBlocksToRead(50);
|
|
// Fast-forward 1 second (100 blocks) since the files start with silence.
|
|
in_file_stereo_->FastForward(100);
|
|
in_file_mono_->FastForward(100);
|
|
|
|
while (1) {
|
|
// Simulate packet loss by setting |packet_loss_| to "true" in
|
|
// |percent_loss| percent of the loops.
|
|
if (percent_loss > 0) {
|
|
if (counter_ == floor((100 / percent_loss) + 0.5)) {
|
|
counter_ = 0;
|
|
channel->set_lost_packet(true);
|
|
} else {
|
|
channel->set_lost_packet(false);
|
|
}
|
|
counter_++;
|
|
}
|
|
|
|
// Add 10 msec to ACM
|
|
if (in_channels == 1) {
|
|
if (in_file_mono_->EndOfFile()) {
|
|
break;
|
|
}
|
|
in_file_mono_->Read10MsData(audio_frame);
|
|
} else {
|
|
if (in_file_stereo_->EndOfFile()) {
|
|
break;
|
|
}
|
|
in_file_stereo_->Read10MsData(audio_frame);
|
|
}
|
|
EXPECT_GE(acm_a_->Add10MsData(audio_frame), 0);
|
|
|
|
// Verify that the received packet size matches the settings.
|
|
rec_size = channel->payload_size();
|
|
if ((0 < rec_size) & (rec_size < 65535)) {
|
|
if (strcmp(send_codec_name_, "opus") == 0) {
|
|
// Opus is a variable rate codec, hence calculate the average packet
|
|
// size, and later make sure the average is in the right range.
|
|
variable_bytes += rec_size;
|
|
variable_packets++;
|
|
} else {
|
|
// For fixed rate codecs, check that packet size is correct.
|
|
if ((rec_size != pack_size_bytes_ * out_channels) &&
|
|
(pack_size_bytes_ < 65535)) {
|
|
error_count++;
|
|
}
|
|
}
|
|
// Verify that the timestamp is updated with expected length
|
|
time_stamp_diff = channel->timestamp_diff();
|
|
if ((counter_ > 10) && (time_stamp_diff != pack_size_samp_)) {
|
|
error_count++;
|
|
}
|
|
}
|
|
|
|
// Run receive side of ACM
|
|
bool muted;
|
|
EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
|
|
ASSERT_FALSE(muted);
|
|
|
|
// Write output speech to file
|
|
out_file_.Write10MsData(
|
|
audio_frame.data(),
|
|
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
|
|
}
|
|
|
|
EXPECT_EQ(0, error_count);
|
|
|
|
// Check that packet size is in the right range for variable rate codecs,
|
|
// such as Opus.
|
|
if (variable_packets > 0) {
|
|
variable_bytes /= variable_packets;
|
|
EXPECT_NEAR(variable_bytes, pack_size_bytes_, 18);
|
|
}
|
|
|
|
if (in_file_mono_->EndOfFile()) {
|
|
in_file_mono_->Rewind();
|
|
}
|
|
if (in_file_stereo_->EndOfFile()) {
|
|
in_file_stereo_->Rewind();
|
|
}
|
|
// Reset in case we ended with a lost packet
|
|
channel->set_lost_packet(false);
|
|
}
|
|
|
|
void TestStereo::OpenOutFile(int16_t test_number) {
|
|
std::string file_name;
|
|
rtc::StringBuilder file_stream;
|
|
file_stream << webrtc::test::OutputPath() << "teststereo_out_" << test_number
|
|
<< ".pcm";
|
|
file_name = file_stream.str();
|
|
out_file_.Open(file_name, 32000, "wb");
|
|
}
|
|
|
|
} // namespace webrtc
|