audio_coding: remove "main" directory

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

Index: webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
deleted file mode 100644
index fdcfdfc22de07096f75e29ced7dbb328a03cb901..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
+++ /dev/null
@@ -1,221 +0,0 @@
-/*
- *  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_receive_test_oldapi.h"
-
-#include <assert.h>
-#include <stdio.h>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-
-namespace webrtc {
-namespace test {
-
-namespace {
-// Returns true if the codec should be registered, otherwise false. Changes
-// the number of channels for the Opus codec to always be 1.
-bool ModifyAndUseThisCodec(CodecInst* codec_param) {
-  if (STR_CASE_CMP(codec_param->plname, "CN") == 0 &&
-      codec_param->plfreq == 48000)
-    return false;  // Skip 48 kHz comfort noise.
-
-  if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0)
-    return false;  // Skip DTFM.
-
-  return true;
-}
-
-// Remaps payload types from ACM's default to those used in the resource file
-// neteq_universal_new.rtp. Returns true if the codec should be registered,
-// otherwise false. The payload types are set as follows (all are mono codecs):
-// PCMu = 0;
-// PCMa = 8;
-// Comfort noise 8 kHz = 13
-// Comfort noise 16 kHz = 98
-// Comfort noise 32 kHz = 99
-// iLBC = 102
-// iSAC wideband = 103
-// iSAC super-wideband = 104
-// AVT/DTMF = 106
-// RED = 117
-// PCM16b 8 kHz = 93
-// PCM16b 16 kHz = 94
-// PCM16b 32 kHz = 95
-// G.722 = 94
-bool RemapPltypeAndUseThisCodec(const char* plname,
-                                int plfreq,
-                                int channels,
-                                int* pltype) {
-  if (channels != 1)
-    return false;  // Don't use non-mono codecs.
-
-  // Re-map pltypes to those used in the NetEq test files.
-  if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) {
-    *pltype = 0;
-  } else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) {
-    *pltype = 8;
-  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) {
-    *pltype = 13;
-  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) {
-    *pltype = 98;
-  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) {
-    *pltype = 99;
-  } else if (STR_CASE_CMP(plname, "ILBC") == 0) {
-    *pltype = 102;
-  } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) {
-    *pltype = 103;
-  } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
-    *pltype = 104;
-  } else if (STR_CASE_CMP(plname, "telephone-event") == 0) {
-    *pltype = 106;
-  } else if (STR_CASE_CMP(plname, "red") == 0) {
-    *pltype = 117;
-  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) {
-    *pltype = 93;
-  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) {
-    *pltype = 94;
-  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
-    *pltype = 95;
-  } else if (STR_CASE_CMP(plname, "G722") == 0) {
-    *pltype = 9;
-  } else {
-    // Don't use any other codecs.
-    return false;
-  }
-  return true;
-}
-}  // namespace
-
-AcmReceiveTestOldApi::AcmReceiveTestOldApi(
-    PacketSource* packet_source,
-    AudioSink* audio_sink,
-    int output_freq_hz,
-    NumOutputChannels exptected_output_channels)
-    : clock_(0),
-      acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
-      packet_source_(packet_source),
-      audio_sink_(audio_sink),
-      output_freq_hz_(output_freq_hz),
-      exptected_output_channels_(exptected_output_channels) {
-}
-
-void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
-  CodecInst my_codec_param;
-  for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
-    ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
-    if (ModifyAndUseThisCodec(&my_codec_param)) {
-      ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
-          << "Couldn't register receive codec.\n";
-    }
-  }
-}
-
-void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
-  CodecInst my_codec_param;
-  for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
-    ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
-    if (!ModifyAndUseThisCodec(&my_codec_param)) {
-      // Skip this codec.
-      continue;
-    }
-
-    if (RemapPltypeAndUseThisCodec(my_codec_param.plname,
-                                   my_codec_param.plfreq,
-                                   my_codec_param.channels,
-                                   &my_codec_param.pltype)) {
-      ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
-          << "Couldn't register receive codec.\n";
-    }
-  }
-}
-
-int AcmReceiveTestOldApi::RegisterExternalReceiveCodec(
-    int rtp_payload_type,
-    AudioDecoder* external_decoder,
-    int sample_rate_hz,
-    int num_channels) {
-  return acm_->RegisterExternalReceiveCodec(rtp_payload_type, external_decoder,
-                                            sample_rate_hz, num_channels);
-}
-
-void AcmReceiveTestOldApi::Run() {
-  for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
-       packet.reset(packet_source_->NextPacket())) {
-    // Pull audio until time to insert packet.
-    while (clock_.TimeInMilliseconds() < packet->time_ms()) {
-      AudioFrame output_frame;
-      EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
-      EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
-      const size_t samples_per_block =
-          static_cast<size_t>(output_freq_hz_ * 10 / 1000);
-      EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
-      if (exptected_output_channels_ != kArbitraryChannels) {
-        if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
-          // Don't check number of channels for PLC output, since each test run
-          // usually starts with a short period of mono PLC before decoding the
-          // first packet.
-        } else {
-          EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
-        }
-      }
-      ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
-      clock_.AdvanceTimeMilliseconds(10);
-      AfterGetAudio();
-    }
-
-    // Insert packet after converting from RTPHeader to WebRtcRTPHeader.
-    WebRtcRTPHeader header;
-    header.header = packet->header();
-    header.frameType = kAudioFrameSpeech;
-    memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
-    EXPECT_EQ(0,
-              acm_->IncomingPacket(
-                  packet->payload(),
-                  static_cast<int32_t>(packet->payload_length_bytes()),
-                  header))
-        << "Failure when inserting packet:" << std::endl
-        << "  PT = " << static_cast<int>(header.header.payloadType) << std::endl
-        << "  TS = " << header.header.timestamp << std::endl
-        << "  SN = " << header.header.sequenceNumber;
-  }
-}
-
-AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
-    PacketSource* packet_source,
-    AudioSink* audio_sink,
-    int output_freq_hz_1,
-    int output_freq_hz_2,
-    int toggle_period_ms,
-    NumOutputChannels exptected_output_channels)
-    : AcmReceiveTestOldApi(packet_source,
-                           audio_sink,
-                           output_freq_hz_1,
-                           exptected_output_channels),
-      output_freq_hz_1_(output_freq_hz_1),
-      output_freq_hz_2_(output_freq_hz_2),
-      toggle_period_ms_(toggle_period_ms),
-      last_toggle_time_ms_(clock_.TimeInMilliseconds()) {
-}
-
-void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
-  if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
-    output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
-                          ? output_freq_hz_2_
-                          : output_freq_hz_1_;
-    last_toggle_time_ms_ = clock_.TimeInMilliseconds();
-  }
-}
-
-}  // namespace test
-}  // namespace webrtc