Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 412 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestAllCodecs.cc b/third_party/libwebrtc/modules/audio_coding/test/TestAllCodecs.cc
new file mode 100644
index 0000000000..b44037d732
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestAllCodecs.cc
@@ -0,0 +1,412 @@
+/*
+ *  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 "modules/audio_coding/test/TestAllCodecs.h"
+
+#include <cstdio>
+#include <limits>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.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".
+
+#define CHECK_ERROR(f)                      \
+  do {                                      \
+    EXPECT_GE(f, 0) << "Error Calling API"; \
+  } while (0)
+
+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(AudioFrameType frame_type,
+                           uint8_t payload_type,
+                           uint32_t timestamp,
+                           const uint8_t* payload_data,
+                           size_t payload_size,
+                           int64_t absolute_capture_timestamp_ms) {
+  RTPHeader rtp_header;
+  int32_t status;
+
+  rtp_header.markerBit = false;
+  rtp_header.ssrc = 0;
+  rtp_header.sequenceNumber = sequence_number_++;
+  rtp_header.payloadType = payload_type;
+  rtp_header.timestamp = timestamp;
+
+  if (frame_type == AudioFrameType::kEmptyFrame) {
+    // Skip this frame.
+    return 0;
+  }
+
+  // Only run mono for all test cases.
+  memcpy(payload_data_, payload_data, payload_size);
+
+  status =
+      receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_header);
+
+  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()
+    : acm_a_(AudioCodingModule::Create(
+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+      acm_b_(AudioCodingModule::Create(
+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+      channel_a_to_b_(NULL),
+      test_count_(0),
+      packet_size_samples_(0),
+      packet_size_bytes_(0) {}
+
+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");
+
+  acm_a_->InitializeReceiver();
+  acm_b_->InitializeReceiver();
+
+  acm_b_->SetReceiveCodecs({{107, {"L16", 8000, 1}},
+                            {108, {"L16", 16000, 1}},
+                            {109, {"L16", 32000, 1}},
+                            {111, {"L16", 8000, 2}},
+                            {112, {"L16", 16000, 2}},
+                            {113, {"L16", 32000, 2}},
+                            {0, {"PCMU", 8000, 1}},
+                            {110, {"PCMU", 8000, 2}},
+                            {8, {"PCMA", 8000, 1}},
+                            {118, {"PCMA", 8000, 2}},
+                            {102, {"ILBC", 8000, 1}},
+                            {9, {"G722", 8000, 1}},
+                            {119, {"G722", 8000, 2}},
+                            {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+                            {13, {"CN", 8000, 1}},
+                            {98, {"CN", 16000, 1}},
+                            {99, {"CN", 32000, 1}}});
+
+  // 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.
+  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();
+#ifdef WEBRTC_CODEC_ILBC
+  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
+  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();
+
+  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();
+
+  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();
+
+  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_);
+
+  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
+  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
+}
+
+// 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) {
+  // 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.
+  int clockrate_hz = sampling_freq_hz;
+  size_t num_channels = 1;
+  if (absl::EqualsIgnoreCase(codec_name, "G722")) {
+    packet_size_samples_ = packet_size / 2;
+    clockrate_hz = sampling_freq_hz / 2;
+  } else if (absl::EqualsIgnoreCase(codec_name, "OPUS")) {
+    packet_size_samples_ = packet_size;
+    num_channels = 2;
+  } 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);
+
+  auto factory = CreateBuiltinAudioEncoderFactory();
+  constexpr int payload_type = 17;
+  SdpAudioFormat format = {codec_name, clockrate_hz, num_channels};
+  format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
+      packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
+  my_acm->SetEncoder(
+      factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
+}
+
+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;
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  infile_a_.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  infile_a_.FastForward(100);
+
+  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.
+    bool muted;
+    CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
+
+    // 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();
+  rtc::StringBuilder test_number_str;
+  test_number_str << test_number;
+  filename += "testallcodecs_out_";
+  filename += test_number_str.str();
+  filename += ".pcm";
+  outfile_b_.Open(filename, 32000, "wb");
+}
+
+}  // namespace webrtc