Removing AudioCoding class, a.k.a the new ACM API

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

Index: webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc b/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
deleted file mode 100644
index 85678cf2209ba22e757be1fe8dcee1eebfc9fe90..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
+++ /dev/null
@@ -1,136 +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.h"
-
-#include <assert.h>
-#include <stdio.h>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/interface/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 {
-
-AcmReceiveTest::AcmReceiveTest(PacketSource* packet_source,
-                               AudioSink* audio_sink,
-                               int output_freq_hz,
-                               NumOutputChannels expected_output_channels)
-    : clock_(0),
-      packet_source_(packet_source),
-      audio_sink_(audio_sink),
-      output_freq_hz_(output_freq_hz),
-      expected_output_channels_(expected_output_channels) {
-  webrtc::AudioCoding::Config config;
-  config.clock = &clock_;
-  config.playout_frequency_hz = output_freq_hz_;
-  acm_.reset(webrtc::AudioCoding::Create(config));
-}
-
-void AcmReceiveTest::RegisterDefaultCodecs() {
-#ifdef WEBRTC_CODEC_OPUS
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kOpus, 120));
-#endif
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
-#endif
-#ifdef WEBRTC_CODEC_ISAC
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
-#endif
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 107));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 108));
-  ASSERT_TRUE(
-      acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 109));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B_2ch, 111));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb_2ch, 112));
-  ASSERT_TRUE(
-      acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 113));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU_2ch, 110));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA_2ch, 118));
-#ifdef WEBRTC_CODEC_ILBC
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
-#endif
-#ifdef WEBRTC_CODEC_G722
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722_2ch, 119));
-#endif
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
-#ifdef WEBRTC_CODEC_RED
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 127));
-#endif
-}
-
-void AcmReceiveTest::RegisterNetEqTestCodecs() {
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
-#ifndef WEBRTC_ANDROID
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
-#endif
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 93));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 94));
-  ASSERT_TRUE(
-      acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 95));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
-  ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 117));
-}
-
-void AcmReceiveTest::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_TRUE(acm_->Get10MsAudio(&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 (expected_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(expected_output_channels_, output_frame.num_channels_);
-        }
-      }
-      ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
-      clock_.AdvanceTimeMilliseconds(10);
-    }
-
-    // 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_TRUE(acm_->InsertPacket(packet->payload(),
-                                   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;
-  }
-}
-
-}  // namespace test
-}  // namespace webrtc