audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc

-/*
- *  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 "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
-
-#include <assert.h>
-#include <stdio.h>
-#include <string.h>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-
-namespace webrtc {
-namespace test {
-
-AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
-                                     int source_rate_hz,
-                                     int test_duration_ms)
-    : clock_(0),
-      acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
-      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_(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_;
-  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
-         AudioFrame::kMaxDataSizeSamples);
-  acm_->RegisterTransportCallback(this);
-}
-
-bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
-                                      int sampling_freq_hz,
-                                      int channels,
-                                      int payload_type,
-                                      int frame_size_samples) {
-  CodecInst codec;
-  RTC_CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec,
-                                           sampling_freq_hz, channels));
-  codec.pltype = payload_type;
-  codec.pacsize = frame_size_samples;
-  codec_registered_ = (acm_->RegisterSendCodec(codec) == 0);
-  input_frame_.num_channels_ = channels;
-  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
-         AudioFrame::kMaxDataSizeSamples);
-  return codec_registered_;
-}
-
-bool AcmSendTestOldApi::RegisterExternalCodec(
-    AudioEncoder* external_speech_encoder) {
-  acm_->RegisterExternalSendCodec(external_speech_encoder);
-  input_frame_.num_channels_ = external_speech_encoder->NumChannels();
-  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
-         AudioFrame::kMaxDataSizeSamples);
-  return codec_registered_ = true;
-}
-
-Packet* AcmSendTestOldApi::NextPacket() {
-  assert(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 NULL;
-  }
-  // 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_.data_));
-    if (input_frame_.num_channels_ > 1) {
-      InputAudioFile::DuplicateInterleaved(input_frame_.data_,
-                                           input_block_size_samples_,
-                                           input_frame_.num_channels_,
-                                           input_frame_.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 NULL;
-}
-
-// This method receives the callback from ACM when a new packet is produced.
-int32_t AcmSendTestOldApi::SendData(
-    FrameType frame_type,
-    uint8_t payload_type,
-    uint32_t timestamp,
-    const uint8_t* payload_data,
-    size_t payload_len_bytes,
-    const RTPFragmentationHeader* fragmentation) {
-  // 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);
-  assert(last_payload_vec_.size() == payload_len_bytes);
-  data_to_send_ = true;
-  return 0;
-}
-
-Packet* AcmSendTestOldApi::CreatePacket() {
-  const size_t kRtpHeaderSize = 12;
-  size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize;
-  uint8_t* packet_memory = new uint8_t[allocated_bytes];
-  // 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());
-  Packet* packet =
-      new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds());
-  assert(packet);
-  assert(packet->valid_header());
-  return packet;
-}
-
-}  // namespace test
-}  // namespace webrtc