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/webrtc@7260 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
index 9cfef3a..94d51b7 100644
--- a/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -47,10 +47,9 @@
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 @@
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 @@
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 @@
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 @@
scoped_ptr<AcmReceiver> receiver_;
CodecInst codecs_[ACMCodecDB::kMaxNumCodecs];
- scoped_ptr<AudioCoding> acm_;
+ scoped_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.
@@ -303,7 +307,7 @@
// 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 @@
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 @@
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 @@
// 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/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
deleted file mode 100644
index ef890ec..0000000
--- a/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 <algorithm> // 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<AcmReceiver> receiver_;
- CodecInst codecs_[ACMCodecDB::kMaxNumCodecs];
- scoped_ptr<AudioCodingModule> 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/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 687c5b8..2212f83 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -2073,13 +2073,6 @@
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 @@
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<uint16_t> AudioCodingImpl::GetNackList(
int round_trip_time_ms) const {
return acm_old_->GetNackList(round_trip_time_ms);
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 6e03e5e..93fd96b 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -390,7 +390,10 @@
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 @@
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 @@
virtual void DisableNack() OVERRIDE;
- virtual bool SetVad(bool enable_dtx,
- bool enable_vad,
- ACMVADMode vad_mode) OVERRIDE;
-
virtual std::vector<uint16_t> GetNackList(
int round_trip_time_ms) const OVERRIDE;
@@ -468,11 +465,8 @@
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<acm2::AudioCodingModuleImpl> acm_old_;
- CodecInst current_send_codec_;
+ int playout_frequency_hz_;
};
} // namespace webrtc
diff --git a/modules/audio_coding/main/interface/audio_coding_module.h b/modules/audio_coding/main/interface/audio_coding_module.h
index 8d73285..389b93f 100644
--- a/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/modules/audio_coding/main/interface/audio_coding_module.h
@@ -1015,14 +1015,6 @@
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 @@
// 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 @@
// 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/modules/modules.gyp b/modules/modules.gyp
index 777523a..79af1ba 100644
--- a/modules/modules.gyp
+++ b/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',
