Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface"
Was reverted by mistake in 7260. Actual culprit was 7258.
BUG=3520
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7272 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 94d51b7..9cfef3a 100644
--- a/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -47,9 +47,10 @@
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));
+ AudioCoding::Config config;
+ config.transport = this;
+ acm_.reset(new AudioCodingImpl(config));
+ receiver_.reset(new AcmReceiver(config.ToOldConfig()));
}
~AcmReceiverTest() {}
@@ -61,10 +62,6 @@
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;
@@ -82,12 +79,12 @@
CodecInst codec;
ACMCodecDB::Codec(codec_id, &codec);
if (timestamp_ == 0) { // This is the first time inserting audio.
- ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
+ ASSERT_TRUE(acm_->RegisterSendCodec(codec_id, codec.pltype));
} else {
- CodecInst current_codec;
- ASSERT_EQ(0, acm_->SendCodec(¤t_codec));
- if (!CodecsEqual(codec, current_codec))
- ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
+ const CodecInst* current_codec = acm_->GetSenderCodecInst();
+ ASSERT_TRUE(current_codec);
+ if (!CodecsEqual(codec, *current_codec))
+ ASSERT_TRUE(acm_->RegisterSendCodec(codec_id, codec.pltype));
}
AudioFrame frame;
// Frame setup according to the codec.
@@ -102,8 +99,7 @@
while (num_bytes == 0) {
frame.timestamp_ = timestamp_;
timestamp_ += frame.samples_per_channel_;
- ASSERT_EQ(0, acm_->Add10MsData(frame));
- num_bytes = acm_->Process();
+ num_bytes = acm_->Add10MsAudio(frame);
ASSERT_GE(num_bytes, 0);
}
ASSERT_TRUE(packet_sent_); // Sanity check.
@@ -151,7 +147,7 @@
scoped_ptr<AcmReceiver> receiver_;
CodecInst codecs_[ACMCodecDB::kMaxNumCodecs];
- scoped_ptr<AudioCodingModule> acm_;
+ scoped_ptr<AudioCoding> acm_;
WebRtcRTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.
@@ -307,7 +303,7 @@
// Register CNG at sender side.
int n = 0;
while (kCngId[n] > 0) {
- ASSERT_EQ(0, acm_->RegisterSendCodec(codecs_[kCngId[n]]));
+ ASSERT_TRUE(acm_->RegisterSendCodec(kCngId[n], codecs_[kCngId[n]].pltype));
++n;
}
@@ -316,7 +312,7 @@
EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec));
// Start with sending DTX.
- ASSERT_EQ(0, acm_->SetVAD(true, true, VADVeryAggr));
+ ASSERT_TRUE(acm_->SetVad(true, true, VADVeryAggr));
packet_sent_ = false;
InsertOnePacketOfSilence(kCodecId[0]); // Enough to test with one codec.
ASSERT_TRUE(packet_sent_);
@@ -330,7 +326,7 @@
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]);
@@ -342,7 +338,7 @@
// 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
new file mode 100644
index 0000000..ef890ec
--- /dev/null
+++ b/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
@@ -0,0 +1,364 @@
+/*
+ * 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 2212f83..687c5b8 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -2073,6 +2073,13 @@
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;
@@ -2151,6 +2158,12 @@
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 93fd96b..6e03e5e 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -390,10 +390,7 @@
class AudioCodingImpl : public AudioCoding {
public:
AudioCodingImpl(const Config& config) {
- AudioCodingModule::Config config_old;
- config_old.id = 0;
- config_old.neteq_config = config.neteq_config;
- config_old.clock = config.clock;
+ AudioCodingModule::Config config_old = config.ToOldConfig();
acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old));
acm_old_->RegisterTransportCallback(config.transport);
acm_old_->RegisterVADCallback(config.vad_callback);
@@ -414,6 +411,8 @@
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;
@@ -449,6 +448,10 @@
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;
@@ -465,8 +468,11 @@
int* sample_rate_hz,
int* channels);
- scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
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_;
};
} // 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 389b93f..8d73285 100644
--- a/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/modules/audio_coding/main/interface/audio_coding_module.h
@@ -1015,6 +1015,14 @@
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;
@@ -1046,6 +1054,9 @@
// 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.
@@ -1139,6 +1150,22 @@
// 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 4086299..1025d01 100644
--- a/modules/modules.gyp
+++ b/modules/modules.gyp
@@ -104,6 +104,7 @@
'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/call_statistics_unittest.cc',
'audio_coding/main/acm2/initial_delay_manager_unittest.cc',
