From c570761288acbc3a075be85e8054b2a61c244376 Mon Sep 17 00:00:00 2001 From: "andresp@webrtc.org" Date: Mon, 22 Sep 2014 13:18:34 +0000 Subject: [PATCH] Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258). Breaks android modules_unittests tests by crashing on AcmReceiverBitExactness.8kHzOutput Was already visible on "git cl try" before submitting on https://webrtc-codereview.appspot.com/23719004/# BUG=3520 R=kwiberg@webrtc.org, henrik.lundin@webrtc.org TBR=kwiberg@webrtc.org, henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7260 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../main/acm2/acm_receiver_unittest.cc | 34 +- .../main/acm2/acm_receiver_unittest_oldapi.cc | 364 ------------------ .../main/acm2/audio_coding_module_impl.cc | 13 - .../main/acm2/audio_coding_module_impl.h | 16 +- .../main/interface/audio_coding_module.h | 27 -- webrtc/modules/modules.gyp | 1 - 6 files changed, 24 insertions(+), 431 deletions(-) delete mode 100644 webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc index 9cfef3a8f..94d51b770 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc +++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc @@ -47,10 +47,9 @@ class AcmReceiverTest : public AudioPacketizationCallback, packet_sent_(false), last_packet_send_timestamp_(timestamp_), last_frame_type_(kFrameEmpty) { - AudioCoding::Config config; - config.transport = this; - acm_.reset(new AudioCodingImpl(config)); - receiver_.reset(new AcmReceiver(config.ToOldConfig())); + AudioCodingModule::Config config; + acm_.reset(new AudioCodingModuleImpl(config)); + receiver_.reset(new AcmReceiver(config)); } ~AcmReceiverTest() {} @@ -62,6 +61,10 @@ class AcmReceiverTest : public AudioPacketizationCallback, ASSERT_EQ(0, ACMCodecDB::Codec(n, &codecs_[n])); } + acm_->InitializeReceiver(); + acm_->InitializeSender(); + acm_->RegisterTransportCallback(this); + rtp_header_.header.sequenceNumber = 0; rtp_header_.header.timestamp = 0; rtp_header_.header.markerBit = false; @@ -79,12 +82,12 @@ class AcmReceiverTest : public AudioPacketizationCallback, CodecInst codec; ACMCodecDB::Codec(codec_id, &codec); if (timestamp_ == 0) { // This is the first time inserting audio. - ASSERT_TRUE(acm_->RegisterSendCodec(codec_id, codec.pltype)); + ASSERT_EQ(0, acm_->RegisterSendCodec(codec)); } else { - const CodecInst* current_codec = acm_->GetSenderCodecInst(); - ASSERT_TRUE(current_codec); - if (!CodecsEqual(codec, *current_codec)) - ASSERT_TRUE(acm_->RegisterSendCodec(codec_id, codec.pltype)); + CodecInst current_codec; + ASSERT_EQ(0, acm_->SendCodec(¤t_codec)); + if (!CodecsEqual(codec, current_codec)) + ASSERT_EQ(0, acm_->RegisterSendCodec(codec)); } AudioFrame frame; // Frame setup according to the codec. @@ -99,7 +102,8 @@ class AcmReceiverTest : public AudioPacketizationCallback, while (num_bytes == 0) { frame.timestamp_ = timestamp_; timestamp_ += frame.samples_per_channel_; - num_bytes = acm_->Add10MsAudio(frame); + ASSERT_EQ(0, acm_->Add10MsData(frame)); + num_bytes = acm_->Process(); ASSERT_GE(num_bytes, 0); } ASSERT_TRUE(packet_sent_); // Sanity check. @@ -147,7 +151,7 @@ class AcmReceiverTest : public AudioPacketizationCallback, scoped_ptr receiver_; CodecInst codecs_[ACMCodecDB::kMaxNumCodecs]; - scoped_ptr acm_; + scoped_ptr acm_; WebRtcRTPHeader rtp_header_; uint32_t timestamp_; bool packet_sent_; // Set when SendData is called reset when inserting audio. @@ -303,7 +307,7 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { // Register CNG at sender side. int n = 0; while (kCngId[n] > 0) { - ASSERT_TRUE(acm_->RegisterSendCodec(kCngId[n], codecs_[kCngId[n]].pltype)); + ASSERT_EQ(0, acm_->RegisterSendCodec(codecs_[kCngId[n]])); ++n; } @@ -312,7 +316,7 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec)); // Start with sending DTX. - ASSERT_TRUE(acm_->SetVad(true, true, VADVeryAggr)); + ASSERT_EQ(0, acm_->SetVAD(true, true, VADVeryAggr)); packet_sent_ = false; InsertOnePacketOfSilence(kCodecId[0]); // Enough to test with one codec. ASSERT_TRUE(packet_sent_); @@ -326,7 +330,7 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { n = 0; while (kCodecId[n] >= 0) { // Loop over codecs. // Set DTX off to send audio payload. - acm_->SetVad(false, false, VADAggr); + acm_->SetVAD(false, false, VADAggr); packet_sent_ = false; InsertOnePacketOfSilence(kCodecId[n]); @@ -338,7 +342,7 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { // Set VAD on to send DTX. Then check if the "Last Audio codec" returns // the expected codec. - acm_->SetVad(true, true, VADAggr); + acm_->SetVAD(true, true, VADAggr); // Do as many encoding until a DTX is sent. while (last_frame_type_ != kAudioFrameCN) { diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc deleted file mode 100644 index ef890ecb3..000000000 --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc +++ /dev/null @@ -1,364 +0,0 @@ -/* - * Copyright (c) 2013 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_receiver.h" - -#include // std::min - -#include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" -#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" -#include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/test/test_suite.h" -#include "webrtc/test/testsupport/fileutils.h" -#include "webrtc/test/testsupport/gtest_disable.h" - -namespace webrtc { - -namespace acm2 { -namespace { - -bool CodecsEqual(const CodecInst& codec_a, const CodecInst& codec_b) { - if (strcmp(codec_a.plname, codec_b.plname) != 0 || - codec_a.plfreq != codec_b.plfreq || - codec_a.pltype != codec_b.pltype || - codec_b.channels != codec_a.channels) - return false; - return true; -} - -} // namespace - -class AcmReceiverTestOldApi : public AudioPacketizationCallback, - public ::testing::Test { - protected: - AcmReceiverTestOldApi() - : timestamp_(0), - packet_sent_(false), - last_packet_send_timestamp_(timestamp_), - last_frame_type_(kFrameEmpty) { - AudioCodingModule::Config config; - acm_.reset(new AudioCodingModuleImpl(config)); - receiver_.reset(new AcmReceiver(config)); - } - - ~AcmReceiverTestOldApi() {} - - virtual void SetUp() OVERRIDE { - ASSERT_TRUE(receiver_.get() != NULL); - ASSERT_TRUE(acm_.get() != NULL); - for (int n = 0; n < ACMCodecDB::kNumCodecs; n++) { - ASSERT_EQ(0, ACMCodecDB::Codec(n, &codecs_[n])); - } - - acm_->InitializeReceiver(); - acm_->InitializeSender(); - acm_->RegisterTransportCallback(this); - - rtp_header_.header.sequenceNumber = 0; - rtp_header_.header.timestamp = 0; - rtp_header_.header.markerBit = false; - rtp_header_.header.ssrc = 0x12345678; // Arbitrary. - rtp_header_.header.numCSRCs = 0; - rtp_header_.header.payloadType = 0; - rtp_header_.frameType = kAudioFrameSpeech; - rtp_header_.type.Audio.isCNG = false; - } - - virtual void TearDown() OVERRIDE { - } - - void InsertOnePacketOfSilence(int codec_id) { - CodecInst codec; - ACMCodecDB::Codec(codec_id, &codec); - if (timestamp_ == 0) { // This is the first time inserting audio. - ASSERT_EQ(0, acm_->RegisterSendCodec(codec)); - } else { - CodecInst current_codec; - ASSERT_EQ(0, acm_->SendCodec(¤t_codec)); - if (!CodecsEqual(codec, current_codec)) - ASSERT_EQ(0, acm_->RegisterSendCodec(codec)); - } - AudioFrame frame; - // Frame setup according to the codec. - frame.sample_rate_hz_ = codec.plfreq; - frame.samples_per_channel_ = codec.plfreq / 100; // 10 ms. - frame.num_channels_ = codec.channels; - memset(frame.data_, 0, frame.samples_per_channel_ * frame.num_channels_ * - sizeof(int16_t)); - int num_bytes = 0; - packet_sent_ = false; - last_packet_send_timestamp_ = timestamp_; - while (num_bytes == 0) { - frame.timestamp_ = timestamp_; - timestamp_ += frame.samples_per_channel_; - ASSERT_EQ(0, acm_->Add10MsData(frame)); - num_bytes = acm_->Process(); - ASSERT_GE(num_bytes, 0); - } - ASSERT_TRUE(packet_sent_); // Sanity check. - } - - // Last element of id should be negative. - void AddSetOfCodecs(const int* id) { - int n = 0; - while (id[n] >= 0) { - ASSERT_EQ(0, receiver_->AddCodec(id[n], codecs_[id[n]].pltype, - codecs_[id[n]].channels, NULL)); - ++n; - } - } - - virtual int SendData( - FrameType frame_type, - uint8_t payload_type, - uint32_t timestamp, - const uint8_t* payload_data, - uint16_t payload_len_bytes, - const RTPFragmentationHeader* fragmentation) OVERRIDE { - if (frame_type == kFrameEmpty) - return 0; - - rtp_header_.header.payloadType = payload_type; - rtp_header_.frameType = frame_type; - if (frame_type == kAudioFrameSpeech) - rtp_header_.type.Audio.isCNG = false; - else - rtp_header_.type.Audio.isCNG = true; - rtp_header_.header.timestamp = timestamp; - - int ret_val = receiver_->InsertPacket(rtp_header_, payload_data, - payload_len_bytes); - if (ret_val < 0) { - assert(false); - return -1; - } - rtp_header_.header.sequenceNumber++; - packet_sent_ = true; - last_frame_type_ = frame_type; - return 0; - } - - scoped_ptr receiver_; - CodecInst codecs_[ACMCodecDB::kMaxNumCodecs]; - scoped_ptr acm_; - WebRtcRTPHeader rtp_header_; - uint32_t timestamp_; - bool packet_sent_; // Set when SendData is called reset when inserting audio. - uint32_t last_packet_send_timestamp_; - FrameType last_frame_type_; -}; - -TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) { - // Add codec. - for (int n = 0; n < ACMCodecDB::kNumCodecs; ++n) { - if (n & 0x1) // Just add codecs with odd index. - EXPECT_EQ(0, receiver_->AddCodec(n, codecs_[n].pltype, - codecs_[n].channels, NULL)); - } - // Get codec and compare. - for (int n = 0; n < ACMCodecDB::kNumCodecs; ++n) { - CodecInst my_codec; - if (n & 0x1) { - // Codecs with odd index should match the reference. - EXPECT_EQ(0, receiver_->DecoderByPayloadType(codecs_[n].pltype, - &my_codec)); - EXPECT_TRUE(CodecsEqual(codecs_[n], my_codec)); - } else { - // Codecs with even index are not registered. - EXPECT_EQ(-1, receiver_->DecoderByPayloadType(codecs_[n].pltype, - &my_codec)); - } - } -} - -TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) { - CodecInst ref_codec; - const int codec_id = ACMCodecDB::kPCMA; - EXPECT_EQ(0, ACMCodecDB::Codec(codec_id, &ref_codec)); - const int payload_type = ref_codec.pltype; - EXPECT_EQ(0, receiver_->AddCodec(codec_id, ref_codec.pltype, - ref_codec.channels, NULL)); - CodecInst test_codec; - EXPECT_EQ(0, receiver_->DecoderByPayloadType(payload_type, &test_codec)); - EXPECT_EQ(true, CodecsEqual(ref_codec, test_codec)); - - // Re-register the same codec with different payload. - ref_codec.pltype = payload_type + 1; - EXPECT_EQ(0, receiver_->AddCodec(codec_id, ref_codec.pltype, - ref_codec.channels, NULL)); - - // Payload type |payload_type| should not exist. - EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &test_codec)); - - // Payload type |payload_type + 1| should exist. - EXPECT_EQ(0, receiver_->DecoderByPayloadType(payload_type + 1, &test_codec)); - EXPECT_TRUE(CodecsEqual(test_codec, ref_codec)); -} - -TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) { - CodecInst codec; - const int codec_id = ACMCodecDB::kPCMA; - EXPECT_EQ(0, ACMCodecDB::Codec(codec_id, &codec)); - const int payload_type = codec.pltype; - EXPECT_EQ(0, receiver_->AddCodec(codec_id, codec.pltype, - codec.channels, NULL)); - - // Remove non-existing codec should not fail. ACM1 legacy. - EXPECT_EQ(0, receiver_->RemoveCodec(payload_type + 1)); - - // Remove an existing codec. - EXPECT_EQ(0, receiver_->RemoveCodec(payload_type)); - - // Ask for the removed codec, must fail. - EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &codec)); -} - -TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) { - const int kCodecId[] = { - ACMCodecDB::kISAC, ACMCodecDB::kISACSWB, ACMCodecDB::kISACFB, - -1 // Terminator. - }; - AddSetOfCodecs(kCodecId); - - AudioFrame frame; - const int kOutSampleRateHz = 8000; // Different than codec sample rate. - int n = 0; - while (kCodecId[n] >= 0) { - const int num_10ms_frames = codecs_[kCodecId[n]].pacsize / - (codecs_[kCodecId[n]].plfreq / 100); - InsertOnePacketOfSilence(kCodecId[n]); - for (int k = 0; k < num_10ms_frames; ++k) { - EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame)); - } - EXPECT_EQ(std::min(32000, codecs_[kCodecId[n]].plfreq), - receiver_->current_sample_rate_hz()); - ++n; - } -} - -// Verify that the playout mode is set correctly. -TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PlayoutMode)) { - receiver_->SetPlayoutMode(voice); - EXPECT_EQ(voice, receiver_->PlayoutMode()); - - receiver_->SetPlayoutMode(streaming); - EXPECT_EQ(streaming, receiver_->PlayoutMode()); - - receiver_->SetPlayoutMode(fax); - EXPECT_EQ(fax, receiver_->PlayoutMode()); - - receiver_->SetPlayoutMode(off); - EXPECT_EQ(off, receiver_->PlayoutMode()); -} - -TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) { - receiver_->EnableVad(); - EXPECT_TRUE(receiver_->vad_enabled()); - - const int id = ACMCodecDB::kPCM16Bwb; - ASSERT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels, - NULL)); - const int kNumPackets = 5; - const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100); - AudioFrame frame; - for (int n = 0; n < kNumPackets; ++n) { - InsertOnePacketOfSilence(id); - for (int k = 0; k < num_10ms_frames; ++k) - ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame)); - } - EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_); - - receiver_->DisableVad(); - EXPECT_FALSE(receiver_->vad_enabled()); - - for (int n = 0; n < kNumPackets; ++n) { - InsertOnePacketOfSilence(id); - for (int k = 0; k < num_10ms_frames; ++k) - ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame)); - } - EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_); -} - -TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(LastAudioCodec)) { - const int kCodecId[] = { - ACMCodecDB::kISAC, ACMCodecDB::kPCMA, ACMCodecDB::kISACSWB, - ACMCodecDB::kPCM16Bswb32kHz, ACMCodecDB::kG722_1C_48, - -1 // Terminator. - }; - AddSetOfCodecs(kCodecId); - - const int kCngId[] = { // Not including full-band. - ACMCodecDB::kCNNB, ACMCodecDB::kCNWB, ACMCodecDB::kCNSWB, - -1 // Terminator. - }; - AddSetOfCodecs(kCngId); - - // Register CNG at sender side. - int n = 0; - while (kCngId[n] > 0) { - ASSERT_EQ(0, acm_->RegisterSendCodec(codecs_[kCngId[n]])); - ++n; - } - - CodecInst codec; - // No audio payload is received. - EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec)); - - // Start with sending DTX. - ASSERT_EQ(0, acm_->SetVAD(true, true, VADVeryAggr)); - packet_sent_ = false; - InsertOnePacketOfSilence(kCodecId[0]); // Enough to test with one codec. - ASSERT_TRUE(packet_sent_); - EXPECT_EQ(kAudioFrameCN, last_frame_type_); - - // Has received, only, DTX. Last Audio codec is undefined. - EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec)); - EXPECT_EQ(-1, receiver_->last_audio_codec_id()); - EXPECT_EQ(-1, receiver_->last_audio_payload_type()); - - n = 0; - while (kCodecId[n] >= 0) { // Loop over codecs. - // Set DTX off to send audio payload. - acm_->SetVAD(false, false, VADAggr); - packet_sent_ = false; - InsertOnePacketOfSilence(kCodecId[n]); - - // Sanity check if Actually an audio payload received, and it should be - // of type "speech." - ASSERT_TRUE(packet_sent_); - ASSERT_EQ(kAudioFrameSpeech, last_frame_type_); - EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id()); - - // Set VAD on to send DTX. Then check if the "Last Audio codec" returns - // the expected codec. - acm_->SetVAD(true, true, VADAggr); - - // Do as many encoding until a DTX is sent. - while (last_frame_type_ != kAudioFrameCN) { - packet_sent_ = false; - InsertOnePacketOfSilence(kCodecId[n]); - ASSERT_TRUE(packet_sent_); - } - EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id()); - EXPECT_EQ(codecs_[kCodecId[n]].pltype, - receiver_->last_audio_payload_type()); - EXPECT_EQ(0, receiver_->LastAudioCodec(&codec)); - EXPECT_TRUE(CodecsEqual(codecs_[kCodecId[n]], codec)); - ++n; - } -} - -} // namespace acm2 - -} // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc index 687c5b810..2212f83c3 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc @@ -2073,13 +2073,6 @@ const AudioEncoder* AudioCodingImpl::GetSenderInfo() const { FATAL() << "Not implemented yet."; } -const CodecInst* AudioCodingImpl::GetSenderCodecInst() { - if (acm_old_->SendCodec(¤t_send_codec_) != 0) { - return NULL; - } - return ¤t_send_codec_; -} - int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) { if (acm_old_->Add10MsData(audio_frame) != 0) { return -1; @@ -2158,12 +2151,6 @@ void AudioCodingImpl::DisableNack() { FATAL() << "Not implemented yet."; } -bool AudioCodingImpl::SetVad(bool enable_dtx, - bool enable_vad, - ACMVADMode vad_mode) { - return acm_old_->SetVAD(enable_dtx, enable_vad, vad_mode) == 0; -} - std::vector AudioCodingImpl::GetNackList( int round_trip_time_ms) const { return acm_old_->GetNackList(round_trip_time_ms); diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h index 6e03e5e61..93fd96bec 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h @@ -390,7 +390,10 @@ class AudioCodingModuleImpl : public AudioCodingModule { class AudioCodingImpl : public AudioCoding { public: AudioCodingImpl(const Config& config) { - AudioCodingModule::Config config_old = config.ToOldConfig(); + AudioCodingModule::Config config_old; + config_old.id = 0; + config_old.neteq_config = config.neteq_config; + config_old.clock = config.clock; acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old)); acm_old_->RegisterTransportCallback(config.transport); acm_old_->RegisterVADCallback(config.vad_callback); @@ -411,8 +414,6 @@ class AudioCodingImpl : public AudioCoding { virtual const AudioEncoder* GetSenderInfo() const OVERRIDE; - virtual const CodecInst* GetSenderCodecInst() OVERRIDE; - virtual int Add10MsAudio(const AudioFrame& audio_frame) OVERRIDE; virtual const ReceiverInfo* GetReceiverInfo() const OVERRIDE; @@ -448,10 +449,6 @@ class AudioCodingImpl : public AudioCoding { virtual void DisableNack() OVERRIDE; - virtual bool SetVad(bool enable_dtx, - bool enable_vad, - ACMVADMode vad_mode) OVERRIDE; - virtual std::vector GetNackList( int round_trip_time_ms) const OVERRIDE; @@ -468,11 +465,8 @@ class AudioCodingImpl : public AudioCoding { int* sample_rate_hz, int* channels); - int playout_frequency_hz_; - // TODO(henrik.lundin): All members below this line are temporary and should - // be removed after refactoring is completed. scoped_ptr acm_old_; - CodecInst current_send_codec_; + int playout_frequency_hz_; }; } // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h index 8d73285a5..389b93fe7 100644 --- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h +++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h @@ -1015,14 +1015,6 @@ class AudioCoding { playout_channels(1), playout_frequency_hz(32000) {} - AudioCodingModule::Config ToOldConfig() const { - AudioCodingModule::Config old_config; - old_config.id = 0; - old_config.neteq_config = neteq_config; - old_config.clock = clock; - return old_config; - } - NetEq::Config neteq_config; Clock* clock; AudioPacketizationCallback* transport; @@ -1054,9 +1046,6 @@ class AudioCoding { // codec that was registered in the latest call to RegisterSendCodec(). virtual const AudioEncoder* GetSenderInfo() const = 0; - // Temporary solution to be used during refactoring. - virtual const CodecInst* GetSenderCodecInst() = 0; - // Adds 10 ms of raw (PCM) audio data to the encoder. If the sampling // frequency of the audio does not match the sampling frequency of the // current encoder, ACM will resample the audio. @@ -1150,22 +1139,6 @@ class AudioCoding { // Disables NACK. virtual void DisableNack() = 0; - - // Temporary solution to be used during refactoring. - // If DTX is enabled and the codec does not have internal DTX/VAD - // WebRtc VAD will be automatically enabled and |enable_vad| is ignored. - // - // If DTX is disabled but VAD is enabled no DTX packets are sent, - // regardless of whether the codec has internal DTX/VAD or not. In this - // case, WebRtc VAD is running to label frames as active/in-active. - // - // NOTE! VAD/DTX is not supported when sending stereo. - // - // Return true if successful, false otherwise. - virtual bool SetVad(bool enable_dtx, - bool enable_vad, - ACMVADMode vad_mode) = 0; - // Returns a list of packets to request retransmission of. // |round_trip_time_ms| is an estimate of the round-trip-time (in // milliseconds). Missing packets which will be decoded sooner than the diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 777523aba..79af1baba 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -104,7 +104,6 @@ 'sources': [ 'audio_coding/main/acm2/acm_opus_unittest.cc', 'audio_coding/main/acm2/acm_receiver_unittest.cc', - 'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc', 'audio_coding/main/acm2/audio_coding_module_unittest.cc', 'audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc', 'audio_coding/main/acm2/call_statistics_unittest.cc',