1 files changed, 174 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_send_test.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_send_test.cc
new file mode 100644
index 0000000000..3e65f94b0d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_send_test.cc
@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2014 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/acm2/acm_send_test.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_encoder.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.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_encode.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
+                                     int source_rate_hz,
+                                     int test_duration_ms)
+    : clock_(0),
+      acm_(webrtc::AudioCodingModule::Create([this] {
+        AudioCodingModule::Config config;
+        config.clock = &clock_;
+        config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+        return config;
+      }())),
+      audio_source_(audio_source),
+      source_rate_hz_(source_rate_hz),
+      input_block_size_samples_(
+          static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),
+      codec_registered_(false),
+      test_duration_ms_(test_duration_ms),
+      frame_type_(AudioFrameType::kAudioFrameSpeech),
+      payload_type_(0),
+      timestamp_(0),
+      sequence_number_(0) {
+  input_frame_.sample_rate_hz_ = source_rate_hz_;
+  input_frame_.num_channels_ = 1;
+  input_frame_.samples_per_channel_ = input_block_size_samples_;
+  RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
+  acm_->RegisterTransportCallback(this);
+}
+
+AcmSendTestOldApi::~AcmSendTestOldApi() = default;
+
+bool AcmSendTestOldApi::RegisterCodec(absl::string_view payload_name,
+                                      int clockrate_hz,
+                                      int num_channels,
+                                      int payload_type,
+                                      int frame_size_samples) {
+  SdpAudioFormat format(payload_name, clockrate_hz, num_channels);
+  if (absl::EqualsIgnoreCase(payload_name, "g722")) {
+    RTC_CHECK_EQ(16000, clockrate_hz);
+    format.clockrate_hz = 8000;
+  } else if (absl::EqualsIgnoreCase(payload_name, "opus")) {
+    RTC_CHECK(num_channels == 1 || num_channels == 2);
+    if (num_channels == 2) {
+      format.parameters["stereo"] = "1";
+    }
+    format.num_channels = 2;
+  }
+  format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
+      frame_size_samples, rtc::CheckedDivExact(clockrate_hz, 1000)));
+  auto factory = CreateBuiltinAudioEncoderFactory();
+  acm_->SetEncoder(
+      factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
+  codec_registered_ = true;
+  input_frame_.num_channels_ = num_channels;
+  RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
+  return codec_registered_;
+}
+
+void AcmSendTestOldApi::RegisterExternalCodec(
+    std::unique_ptr<AudioEncoder> external_speech_encoder) {
+  input_frame_.num_channels_ = external_speech_encoder->NumChannels();
+  acm_->SetEncoder(std::move(external_speech_encoder));
+  RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
+  codec_registered_ = true;
+}
+
+std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
+  RTC_DCHECK(codec_registered_);
+  if (filter_.test(static_cast<size_t>(payload_type_))) {
+    // This payload type should be filtered out. Since the payload type is the
+    // same throughout the whole test run, no packet at all will be delivered.
+    // We can just as well signal that the test is over by returning NULL.
+    return nullptr;
+  }
+  // Insert audio and process until one packet is produced.
+  while (clock_.TimeInMilliseconds() < test_duration_ms_) {
+    clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
+    RTC_CHECK(audio_source_->Read(
+        input_block_size_samples_ * input_frame_.num_channels_,
+        input_frame_.mutable_data()));
+    data_to_send_ = false;
+    RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
+    input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
+    if (data_to_send_) {
+      // Encoded packet received.
+      return CreatePacket();
+    }
+  }
+  // Test ended.
+  return nullptr;
+}
+
+// This method receives the callback from ACM when a new packet is produced.
+int32_t AcmSendTestOldApi::SendData(AudioFrameType frame_type,
+                                    uint8_t payload_type,
+                                    uint32_t timestamp,
+                                    const uint8_t* payload_data,
+                                    size_t payload_len_bytes,
+                                    int64_t absolute_capture_timestamp_ms) {
+  // Store the packet locally.
+  frame_type_ = frame_type;
+  payload_type_ = payload_type;
+  timestamp_ = timestamp;
+  last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
+  RTC_DCHECK_EQ(last_payload_vec_.size(), payload_len_bytes);
+  data_to_send_ = true;
+  return 0;
+}
+
+std::unique_ptr<Packet> AcmSendTestOldApi::CreatePacket() {
+  const size_t kRtpHeaderSize = 12;
+  rtc::CopyOnWriteBuffer packet_buffer(last_payload_vec_.size() +
+                                       kRtpHeaderSize);
+  uint8_t* packet_memory = packet_buffer.MutableData();
+  // Populate the header bytes.
+  packet_memory[0] = 0x80;
+  packet_memory[1] = static_cast<uint8_t>(payload_type_);
+  packet_memory[2] = (sequence_number_ >> 8) & 0xFF;
+  packet_memory[3] = (sequence_number_)&0xFF;
+  packet_memory[4] = (timestamp_ >> 24) & 0xFF;
+  packet_memory[5] = (timestamp_ >> 16) & 0xFF;
+  packet_memory[6] = (timestamp_ >> 8) & 0xFF;
+  packet_memory[7] = timestamp_ & 0xFF;
+  // Set SSRC to 0x12345678.
+  packet_memory[8] = 0x12;
+  packet_memory[9] = 0x34;
+  packet_memory[10] = 0x56;
+  packet_memory[11] = 0x78;
+
+  ++sequence_number_;
+
+  // Copy the payload data.
+  memcpy(packet_memory + kRtpHeaderSize, &last_payload_vec_[0],
+         last_payload_vec_.size());
+  auto packet = std::make_unique<Packet>(std::move(packet_buffer),
+                                         clock_.TimeInMilliseconds());
+  RTC_DCHECK(packet);
+  RTC_DCHECK(packet->valid_header());
+  return packet;
+}
+
+}  // namespace test
+}  // namespace webrtc