audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
diff --git a/webrtc/modules/audio_coding/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
new file mode 100644
index 0000000..21ce7c1
--- /dev/null
+++ b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
@@ -0,0 +1,485 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
+
+#include <cstdio>
+#include <limits>
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/typedefs.h"
+
+// Description of the test:
+// In this test we set up a one-way communication channel from a participant
+// called "a" to a participant called "b".
+// a -> channel_a_to_b -> b
+//
+// The test loops through all available mono codecs, encode at "a" sends over
+// the channel, and decodes at "b".
+
+namespace {
+const size_t kVariableSize = std::numeric_limits<size_t>::max();
+}
+
+namespace webrtc {
+
+// Class for simulating packet handling.
+TestPack::TestPack()
+ : receiver_acm_(NULL),
+ sequence_number_(0),
+ timestamp_diff_(0),
+ last_in_timestamp_(0),
+ total_bytes_(0),
+ payload_size_(0) {
+}
+
+TestPack::~TestPack() {
+}
+
+void TestPack::RegisterReceiverACM(AudioCodingModule* acm) {
+ receiver_acm_ = acm;
+ return;
+}
+
+int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type,
+ uint32_t timestamp, const uint8_t* payload_data,
+ size_t payload_size,
+ const RTPFragmentationHeader* fragmentation) {
+ WebRtcRTPHeader rtp_info;
+ int32_t status;
+
+ rtp_info.header.markerBit = false;
+ rtp_info.header.ssrc = 0;
+ rtp_info.header.sequenceNumber = sequence_number_++;
+ rtp_info.header.payloadType = payload_type;
+ rtp_info.header.timestamp = timestamp;
+ if (frame_type == kAudioFrameCN) {
+ rtp_info.type.Audio.isCNG = true;
+ } else {
+ rtp_info.type.Audio.isCNG = false;
+ }
+ if (frame_type == kEmptyFrame) {
+ // Skip this frame.
+ return 0;
+ }
+
+ // Only run mono for all test cases.
+ rtp_info.type.Audio.channel = 1;
+ memcpy(payload_data_, payload_data, payload_size);
+
+ status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);
+
+ payload_size_ = payload_size;
+ timestamp_diff_ = timestamp - last_in_timestamp_;
+ last_in_timestamp_ = timestamp;
+ total_bytes_ += payload_size;
+ return status;
+}
+
+size_t TestPack::payload_size() {
+ return payload_size_;
+}
+
+uint32_t TestPack::timestamp_diff() {
+ return timestamp_diff_;
+}
+
+void TestPack::reset_payload_size() {
+ payload_size_ = 0;
+}
+
+TestAllCodecs::TestAllCodecs(int test_mode)
+ : acm_a_(AudioCodingModule::Create(0)),
+ acm_b_(AudioCodingModule::Create(1)),
+ channel_a_to_b_(NULL),
+ test_count_(0),
+ packet_size_samples_(0),
+ packet_size_bytes_(0) {
+ // test_mode = 0 for silent test (auto test)
+ test_mode_ = test_mode;
+}
+
+TestAllCodecs::~TestAllCodecs() {
+ if (channel_a_to_b_ != NULL) {
+ delete channel_a_to_b_;
+ channel_a_to_b_ = NULL;
+ }
+}
+
+void TestAllCodecs::Perform() {
+ const std::string file_name = webrtc::test::ResourcePath(
+ "audio_coding/testfile32kHz", "pcm");
+ infile_a_.Open(file_name, 32000, "rb");
+
+ if (test_mode_ == 0) {
+ WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+ "---------- TestAllCodecs ----------");
+ }
+
+ acm_a_->InitializeReceiver();
+ acm_b_->InitializeReceiver();
+
+ uint8_t num_encoders = acm_a_->NumberOfCodecs();
+ CodecInst my_codec_param;
+ for (uint8_t n = 0; n < num_encoders; n++) {
+ acm_b_->Codec(n, &my_codec_param);
+ if (!strcmp(my_codec_param.plname, "opus")) {
+ my_codec_param.channels = 1;
+ }
+ acm_b_->RegisterReceiveCodec(my_codec_param);
+ }
+
+ // Create and connect the channel
+ channel_a_to_b_ = new TestPack;
+ acm_a_->RegisterTransportCallback(channel_a_to_b_);
+ channel_a_to_b_->RegisterReceiverACM(acm_b_.get());
+
+ // All codecs are tested for all allowed sampling frequencies, rates and
+ // packet sizes.
+#ifdef WEBRTC_CODEC_G722
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ char codec_g722[] = "G722";
+ RegisterSendCodec('A', codec_g722, 16000, 64000, 160, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_g722, 16000, 64000, 320, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_g722, 16000, 64000, 480, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_g722, 16000, 64000, 640, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_g722, 16000, 64000, 800, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0);
+ Run(channel_a_to_b_);
+ outfile_b_.Close();
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ char codec_ilbc[] = "ILBC";
+ RegisterSendCodec('A', codec_ilbc, 8000, 13300, 240, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_ilbc, 8000, 13300, 480, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_ilbc, 8000, 15200, 160, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_ilbc, 8000, 15200, 320, 0);
+ Run(channel_a_to_b_);
+ outfile_b_.Close();
+#endif
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ char codec_isac[] = "ISAC";
+ RegisterSendCodec('A', codec_isac, 16000, -1, 480, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_isac, 16000, -1, 960, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_isac, 16000, 15000, 480, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_isac, 16000, 32000, 960, kVariableSize);
+ Run(channel_a_to_b_);
+ outfile_b_.Close();
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_isac, 32000, 56000, 960, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_isac, 32000, 37000, 960, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_isac, 32000, 32000, 960, kVariableSize);
+ Run(channel_a_to_b_);
+ outfile_b_.Close();
+#endif
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ char codec_l16[] = "L16";
+ RegisterSendCodec('A', codec_l16, 8000, 128000, 80, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_l16, 8000, 128000, 160, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_l16, 8000, 128000, 240, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_l16, 8000, 128000, 320, 0);
+ Run(channel_a_to_b_);
+ outfile_b_.Close();
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ RegisterSendCodec('A', codec_l16, 16000, 256000, 160, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_l16, 16000, 256000, 320, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_l16, 16000, 256000, 480, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_l16, 16000, 256000, 640, 0);
+ Run(channel_a_to_b_);
+ outfile_b_.Close();
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ RegisterSendCodec('A', codec_l16, 32000, 512000, 320, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0);
+ Run(channel_a_to_b_);
+ outfile_b_.Close();
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ char codec_pcma[] = "PCMA";
+ RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, 0);
+ Run(channel_a_to_b_);
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ char codec_pcmu[] = "PCMU";
+ RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, 0);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_pcmu, 8000, 64000, 480, 0);
+ Run(channel_a_to_b_);
+ outfile_b_.Close();
+#ifdef WEBRTC_CODEC_OPUS
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
+ test_count_++;
+ OpenOutFile(test_count_);
+ char codec_opus[] = "OPUS";
+ RegisterSendCodec('A', codec_opus, 48000, 6000, 480, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 20000, 480*2, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 32000, 480*4, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 48000, 480, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 64000, 480*4, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 96000, 480*6, kVariableSize);
+ Run(channel_a_to_b_);
+ RegisterSendCodec('A', codec_opus, 48000, 500000, 480*2, kVariableSize);
+ Run(channel_a_to_b_);
+ outfile_b_.Close();
+#endif
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+
+ /* Print out all codecs that were not tested in the run */
+ printf("The following codecs was not included in the test:\n");
+#ifndef WEBRTC_CODEC_G722
+ printf(" G.722\n");
+#endif
+#ifndef WEBRTC_CODEC_ILBC
+ printf(" iLBC\n");
+#endif
+#ifndef WEBRTC_CODEC_ISAC
+ printf(" ISAC float\n");
+#endif
+#ifndef WEBRTC_CODEC_ISACFX
+ printf(" ISAC fix\n");
+#endif
+
+ printf("\nTo complete the test, listen to the %d number of output files.\n",
+ test_count_);
+ }
+}
+
+// Register Codec to use in the test
+//
+// Input: side - which ACM to use, 'A' or 'B'
+// codec_name - name to use when register the codec
+// sampling_freq_hz - sampling frequency in Herz
+// rate - bitrate in bytes
+// packet_size - packet size in samples
+// extra_byte - if extra bytes needed compared to the bitrate
+// used when registering, can be an internal header
+// set to kVariableSize if the codec is a variable
+// rate codec
+void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
+ int32_t sampling_freq_hz, int rate,
+ int packet_size, size_t extra_byte) {
+ if (test_mode_ != 0) {
+ // Print out codec and settings.
+ printf("codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name,
+ sampling_freq_hz, rate, packet_size);
+ }
+
+ // Store packet-size in samples, used to validate the received packet.
+ // If G.722, store half the size to compensate for the timestamp bug in the
+ // RFC for G.722.
+ // If iSAC runs in adaptive mode, packet size in samples can change on the
+ // fly, so we exclude this test by setting |packet_size_samples_| to -1.
+ if (!strcmp(codec_name, "G722")) {
+ packet_size_samples_ = packet_size / 2;
+ } else if (!strcmp(codec_name, "ISAC") && (rate == -1)) {
+ packet_size_samples_ = -1;
+ } else {
+ packet_size_samples_ = packet_size;
+ }
+
+ // Store the expected packet size in bytes, used to validate the received
+ // packet. If variable rate codec (extra_byte == -1), set to -1.
+ if (extra_byte != kVariableSize) {
+ // Add 0.875 to always round up to a whole byte
+ packet_size_bytes_ = static_cast<size_t>(
+ static_cast<float>(packet_size * rate) /
+ static_cast<float>(sampling_freq_hz * 8) + 0.875) + extra_byte;
+ } else {
+ // Packets will have a variable size.
+ packet_size_bytes_ = kVariableSize;
+ }
+
+ // 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);
+
+ // Get all codec parameters before registering
+ CodecInst my_codec_param;
+ CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param,
+ sampling_freq_hz, 1));
+ my_codec_param.rate = rate;
+ my_codec_param.pacsize = packet_size;
+ CHECK_ERROR(my_acm->RegisterSendCodec(my_codec_param));
+}
+
+void TestAllCodecs::Run(TestPack* channel) {
+ AudioFrame audio_frame;
+
+ int32_t out_freq_hz = outfile_b_.SamplingFrequency();
+ size_t receive_size;
+ uint32_t timestamp_diff;
+ channel->reset_payload_size();
+ int error_count = 0;
+
+ int counter = 0;
+ while (!infile_a_.EndOfFile()) {
+ // Add 10 msec to ACM.
+ infile_a_.Read10MsData(audio_frame);
+ CHECK_ERROR(acm_a_->Add10MsData(audio_frame));
+
+ // Verify that the received packet size matches the settings.
+ receive_size = channel->payload_size();
+ if (receive_size) {
+ if ((receive_size != packet_size_bytes_) &&
+ (packet_size_bytes_ != kVariableSize)) {
+ error_count++;
+ }
+
+ // Verify that the timestamp is updated with expected length. The counter
+ // is used to avoid problems when switching codec or frame size in the
+ // test.
+ timestamp_diff = channel->timestamp_diff();
+ if ((counter > 10) &&
+ (static_cast<int>(timestamp_diff) != packet_size_samples_) &&
+ (packet_size_samples_ > -1))
+ error_count++;
+ }
+
+ // Run received side of ACM.
+ CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame));
+
+ // Write output speech to file.
+ outfile_b_.Write10MsData(audio_frame.data_,
+ audio_frame.samples_per_channel_);
+
+ // Update loop counter
+ counter++;
+ }
+
+ EXPECT_EQ(0, error_count);
+
+ if (infile_a_.EndOfFile()) {
+ infile_a_.Rewind();
+ }
+}
+
+void TestAllCodecs::OpenOutFile(int test_number) {
+ std::string filename = webrtc::test::OutputPath();
+ std::ostringstream test_number_str;
+ test_number_str << test_number;
+ filename += "testallcodecs_out_";
+ filename += test_number_str.str();
+ filename += ".pcm";
+ outfile_b_.Open(filename, 32000, "wb");
+}
+
+void TestAllCodecs::DisplaySendReceiveCodec() {
+ CodecInst my_codec_param;
+ printf("%s -> ", acm_a_->SendCodec()->plname);
+ acm_b_->ReceiveCodec(&my_codec_param);
+ printf("%s\n", my_codec_param.plname);
+}
+
+} // namespace webrtc