Break apart AudioCodingModule and AcmReceiver
This change makes AudioCodingModule a pure sender and AcmReceiver a pure
receiver.
The Config struct is in practice no longer used by AudioCodingModule,
so a new definition is included in AcmReceiver. The old definition
remains in AudioCodingModule while downstream clients are being
updated.
Bug: webrtc:14867
Change-Id: If0d0b4214c5aa278cf6c85c5b62c6da644de20e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291533
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39244}
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 04ccc98..5b7ff7b 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -65,13 +65,13 @@
constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0;
constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
-AudioCodingModule::Config AcmConfig(
+acm2::AcmReceiver::Config AcmConfig(
NetEqFactory* neteq_factory,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
absl::optional<AudioCodecPairId> codec_pair_id,
size_t jitter_buffer_max_packets,
bool jitter_buffer_fast_playout) {
- AudioCodingModule::Config acm_config;
+ acm2::AcmReceiver::Config acm_config;
acm_config.neteq_factory = neteq_factory;
acm_config.decoder_factory = decoder_factory;
acm_config.neteq_config.codec_pair_id = codec_pair_id;
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index d00688f..3e734aa 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -472,7 +472,7 @@
encoder_queue_(task_queue_factory->CreateTaskQueue(
"AudioEncoder",
TaskQueueFactory::Priority::NORMAL)) {
- audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config()));
+ audio_coding_ = AudioCodingModule::Create();
RtpRtcpInterface::Configuration configuration;
configuration.bandwidth_callback = rtcp_observer_.get();
diff --git a/audio/voip/audio_egress.cc b/audio/voip/audio_egress.cc
index 1162824..ae98d91 100644
--- a/audio/voip/audio_egress.cc
+++ b/audio/voip/audio_egress.cc
@@ -22,7 +22,7 @@
TaskQueueFactory* task_queue_factory)
: rtp_rtcp_(rtp_rtcp),
rtp_sender_audio_(clock, rtp_rtcp_->RtpSender()),
- audio_coding_(AudioCodingModule::Create(AudioCodingModule::Config())),
+ audio_coding_(AudioCodingModule::Create()),
encoder_queue_(task_queue_factory->CreateTaskQueue(
"AudioEncoder",
TaskQueueFactory::Priority::NORMAL)) {
diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index 0f9c9cc..26191c2 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -29,9 +29,9 @@
namespace {
-AudioCodingModule::Config CreateAcmConfig(
+acm2::AcmReceiver::Config CreateAcmConfig(
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
- AudioCodingModule::Config acm_config;
+ acm2::AcmReceiver::Config acm_config;
acm_config.neteq_config.enable_muted_state = true;
acm_config.decoder_factory = decoder_factory;
return acm_config;
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index dddc3ed..fc07126 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1082,8 +1082,6 @@
"test/TestVADDTX.cc",
"test/TestVADDTX.h",
"test/Tester.cc",
- "test/TwoWayCommunication.cc",
- "test/TwoWayCommunication.h",
"test/target_delay_unittest.cc",
]
deps = [
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 8bc76cd..66f6255 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -25,10 +25,10 @@
namespace test {
namespace {
-AudioCodingModule::Config MakeAcmConfig(
- Clock* clock,
+acm2::AcmReceiver::Config MakeAcmConfig(
+ Clock& clock,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
- AudioCodingModule::Config config;
+ acm2::AcmReceiver::Config config;
config.clock = clock;
config.decoder_factory = std::move(decoder_factory);
return config;
@@ -42,8 +42,8 @@
NumOutputChannels exptected_output_channels,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
: clock_(0),
- acm_(webrtc::AudioCodingModule::Create(
- MakeAcmConfig(&clock_, std::move(decoder_factory)))),
+ acm_receiver_(std::make_unique<acm2::AcmReceiver>(
+ MakeAcmConfig(clock_, std::move(decoder_factory)))),
packet_source_(packet_source),
audio_sink_(audio_sink),
output_freq_hz_(output_freq_hz),
@@ -52,43 +52,43 @@
AcmReceiveTestOldApi::~AcmReceiveTestOldApi() = default;
void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
- acm_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {111, {"L16", 8000, 2}},
- {112, {"L16", 16000, 2}},
- {113, {"L16", 32000, 2}},
- {0, {"PCMU", 8000, 1}},
- {110, {"PCMU", 8000, 2}},
- {8, {"PCMA", 8000, 1}},
- {118, {"PCMA", 8000, 2}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {119, {"G722", 8000, 2}},
- {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ acm_receiver_->SetCodecs({{103, {"ISAC", 16000, 1}},
+ {104, {"ISAC", 32000, 1}},
+ {107, {"L16", 8000, 1}},
+ {108, {"L16", 16000, 1}},
+ {109, {"L16", 32000, 1}},
+ {111, {"L16", 8000, 2}},
+ {112, {"L16", 16000, 2}},
+ {113, {"L16", 32000, 2}},
+ {0, {"PCMU", 8000, 1}},
+ {110, {"PCMU", 8000, 2}},
+ {8, {"PCMA", 8000, 1}},
+ {118, {"PCMA", 8000, 2}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {119, {"G722", 8000, 2}},
+ {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
}
// Remaps payload types from ACM's default to those used in the resource file
// neteq_universal_new.rtp.
void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
- acm_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {93, {"L16", 8000, 1}},
- {94, {"L16", 16000, 1}},
- {95, {"L16", 32000, 1}},
- {0, {"PCMU", 8000, 1}},
- {8, {"PCMA", 8000, 1}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {120, {"OPUS", 48000, 2}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ acm_receiver_->SetCodecs({{103, {"ISAC", 16000, 1}},
+ {104, {"ISAC", 32000, 1}},
+ {93, {"L16", 8000, 1}},
+ {94, {"L16", 16000, 1}},
+ {95, {"L16", 32000, 1}},
+ {0, {"PCMU", 8000, 1}},
+ {8, {"PCMA", 8000, 1}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {120, {"OPUS", 48000, 2}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
}
void AcmReceiveTestOldApi::Run() {
@@ -98,8 +98,8 @@
while (clock_.TimeInMilliseconds() < packet->time_ms()) {
AudioFrame output_frame;
bool muted;
- EXPECT_EQ(0,
- acm_->PlayoutData10Ms(output_freq_hz_, &output_frame, &muted));
+ EXPECT_EQ(
+ 0, acm_receiver_->GetAudio(output_freq_hz_, &output_frame, &muted));
ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
ASSERT_FALSE(muted);
const size_t samples_per_block =
@@ -119,10 +119,10 @@
AfterGetAudio();
}
- EXPECT_EQ(0, acm_->IncomingPacket(
- packet->payload(),
- static_cast<int32_t>(packet->payload_length_bytes()),
- packet->header()))
+ EXPECT_EQ(0, acm_receiver_->InsertPacket(
+ packet->header(),
+ rtc::ArrayView<const uint8_t>(
+ packet->payload(), packet->payload_length_bytes())))
<< "Failure when inserting packet:" << std::endl
<< " PT = " << static_cast<int>(packet->header().payloadType)
<< std::endl
diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h
index 2095ef9..d0195dd 100644
--- a/modules/audio_coding/acm2/acm_receive_test.h
+++ b/modules/audio_coding/acm2/acm_receive_test.h
@@ -18,6 +18,7 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/scoped_refptr.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
@@ -57,14 +58,12 @@
// Runs the test and returns true if successful.
void Run();
- AudioCodingModule* get_acm() { return acm_.get(); }
-
protected:
// Method is called after each block of output audio is received from ACM.
virtual void AfterGetAudio() {}
SimulatedClock clock_;
- std::unique_ptr<AudioCodingModule> acm_;
+ std::unique_ptr<acm2::AcmReceiver> acm_receiver_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
int output_freq_hz_;
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index a8fded6..a77e472 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -56,11 +56,6 @@
neteq_config.enable_post_decode_vad = true;
}
-AcmReceiver::Config::Config(const AudioCodingModule::Config& acm_config)
- : neteq_config(acm_config.neteq_config),
- clock(*acm_config.clock),
- decoder_factory(acm_config.decoder_factory) {}
-
AcmReceiver::Config::Config(const Config&) = default;
AcmReceiver::Config::~Config() = default;
@@ -76,9 +71,6 @@
sizeof(int16_t) * AudioFrame::kMaxDataSizeSamples);
}
-AcmReceiver::AcmReceiver(const AudioCodingModule::Config& acm_config)
- : AcmReceiver(Config(acm_config)) {}
-
AcmReceiver::~AcmReceiver() = default;
int AcmReceiver::SetMinimumDelay(int delay_ms) {
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 25105ec..820150a 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -28,7 +28,6 @@
#include "api/neteq/neteq_factory.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
-#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
@@ -46,7 +45,6 @@
struct Config {
explicit Config(
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr);
- explicit Config(const AudioCodingModule::Config& acm_config);
Config(const Config&);
~Config();
@@ -58,9 +56,6 @@
// Constructor of the class
explicit AcmReceiver(const Config& config);
- // Deprecated constructor.
- // TODO(webrtc:14867): Remove when downstream projects are ready.
- explicit AcmReceiver(const AudioCodingModule::Config& acm_config);
// Destructor of the class.
~AcmReceiver();
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 6dd44b6..a5095f0 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -44,11 +44,10 @@
~AcmReceiverTestOldApi() {}
void SetUp() override {
- acm_.reset(AudioCodingModule::Create(config_));
+ acm_ = AudioCodingModule::Create();
receiver_.reset(new AcmReceiver(config_));
ASSERT_TRUE(receiver_.get() != NULL);
ASSERT_TRUE(acm_.get() != NULL);
- acm_->InitializeReceiver();
acm_->RegisterTransportCallback(this);
rtp_header_.sequenceNumber = 0;
@@ -135,7 +134,7 @@
CreateBuiltinAudioEncoderFactory();
const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_ =
CreateBuiltinAudioDecoderFactory();
- AudioCodingModule::Config config_;
+ acm2::AcmReceiver::Config config_;
std::unique_ptr<AcmReceiver> receiver_;
std::unique_ptr<AudioCodingModule> acm_;
RTPHeader rtp_header_;
@@ -383,6 +382,24 @@
EXPECT_EQ(0, stats.decoded_muted_output);
}
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyOutputFrame DISABLED_VerifyOutputFrame
+#else
+#define MAYBE_VerifyOutputFrame VerifyOutputFrame
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_VerifyOutputFrame) {
+ AudioFrame audio_frame;
+ const int kSampleRateHz = 32000;
+ bool muted;
+ EXPECT_EQ(0, receiver_->GetAudio(kSampleRateHz, &audio_frame, &muted));
+ ASSERT_FALSE(muted);
+ EXPECT_EQ(0u, audio_frame.timestamp_);
+ EXPECT_GT(audio_frame.num_channels_, 0u);
+ EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
+ audio_frame.samples_per_channel_);
+ EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
+}
+
// Insert some packets and pull audio. Check statistics are valid. Then,
// simulate packet loss and check if PLC and PLC-to-CNG statistics are
// correctly updated.
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index 3e65f94..fddaa87 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -32,12 +32,7 @@
int source_rate_hz,
int test_duration_ms)
: clock_(0),
- acm_(webrtc::AudioCodingModule::Create([this] {
- AudioCodingModule::Config config;
- config.clock = &clock_;
- config.decoder_factory = CreateBuiltinAudioDecoderFactory();
- return config;
- }())),
+ acm_(webrtc::AudioCodingModule::Create()),
audio_source_(audio_source),
source_rate_hz_(source_rate_hz),
input_block_size_samples_(
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 4367ab0..9618ad2 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -16,7 +16,6 @@
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "api/array_view.h"
-#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/acm2/acm_remixing.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/include/module_common_types.h"
@@ -41,7 +40,7 @@
class AudioCodingModuleImpl final : public AudioCodingModule {
public:
- explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
+ explicit AudioCodingModuleImpl();
~AudioCodingModuleImpl() override;
/////////////////////////////////////////
@@ -66,31 +65,9 @@
int SetPacketLossRate(int loss_rate) override;
/////////////////////////////////////////
- // Receiver
- //
-
- // Initialize receiver, resets codec database etc.
- int InitializeReceiver() override;
-
- void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
-
- // Incoming packet from network parsed and ready for decode.
- int IncomingPacket(const uint8_t* incoming_payload,
- const size_t payload_length,
- const RTPHeader& rtp_info) override;
-
- // Get 10 milliseconds of raw audio data to play out, and
- // automatic resample to the requested frequency if > 0.
- int PlayoutData10Ms(int desired_freq_hz,
- AudioFrame* audio_frame,
- bool* muted) override;
-
- /////////////////////////////////////////
// Statistics
//
- int GetNetworkStatistics(NetworkStatistics* statistics) override;
-
ANAStats GetANAStats() const override;
int GetTargetBitrate() const override;
@@ -134,8 +111,6 @@
absl::optional<int64_t> absolute_capture_timestamp_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
- int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
-
bool HaveValidEncoder(absl::string_view caller_name) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
@@ -163,7 +138,6 @@
uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_mutex_);
uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_mutex_);
acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_mutex_);
- acm2::AcmReceiver receiver_; // AcmReceiver has it's own internal lock.
ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_mutex_);
// Current encoder stack, provided by a call to RegisterEncoder.
@@ -172,8 +146,6 @@
// This is to keep track of CN instances where we can send DTMFs.
uint8_t previous_pltype_ RTC_GUARDED_BY(acm_mutex_);
- bool receiver_initialized_ RTC_GUARDED_BY(acm_mutex_);
-
AudioFrame preprocess_frame_ RTC_GUARDED_BY(acm_mutex_);
bool first_10ms_data_ RTC_GUARDED_BY(acm_mutex_);
@@ -206,23 +178,17 @@
}
}
-AudioCodingModuleImpl::AudioCodingModuleImpl(
- const AudioCodingModule::Config& config)
+AudioCodingModuleImpl::AudioCodingModuleImpl()
: expected_codec_ts_(0xD87F3F9F),
expected_in_ts_(0xD87F3F9F),
- receiver_(config),
bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
encoder_stack_(nullptr),
previous_pltype_(255),
- receiver_initialized_(false),
first_10ms_data_(false),
first_frame_(true),
packetization_callback_(NULL),
codec_histogram_bins_log_(),
number_of_consecutive_empty_packets_(0) {
- if (InitializeReceiverSafe() < 0) {
- RTC_LOG(LS_ERROR) << "Cannot initialize receiver";
- }
RTC_LOG(LS_INFO) << "Created";
}
@@ -529,67 +495,9 @@
}
/////////////////////////////////////////
-// Receiver
-//
-
-int AudioCodingModuleImpl::InitializeReceiver() {
- MutexLock lock(&acm_mutex_);
- return InitializeReceiverSafe();
-}
-
-// Initialize receiver, resets codec database etc.
-int AudioCodingModuleImpl::InitializeReceiverSafe() {
- // If the receiver is already initialized then we want to destroy any
- // existing decoders. After a call to this function, we should have a clean
- // start-up.
- if (receiver_initialized_)
- receiver_.RemoveAllCodecs();
- receiver_.FlushBuffers();
-
- receiver_initialized_ = true;
- return 0;
-}
-
-void AudioCodingModuleImpl::SetReceiveCodecs(
- const std::map<int, SdpAudioFormat>& codecs) {
- MutexLock lock(&acm_mutex_);
- receiver_.SetCodecs(codecs);
-}
-
-// Incoming packet from network parsed and ready for decode.
-int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
- const size_t payload_length,
- const RTPHeader& rtp_header) {
- RTC_DCHECK_EQ(payload_length == 0, incoming_payload == nullptr);
- return receiver_.InsertPacket(
- rtp_header,
- rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
-}
-
-// Get 10 milliseconds of raw audio data to play out.
-// Automatic resample to the requested frequency.
-int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
- AudioFrame* audio_frame,
- bool* muted) {
- // GetAudio always returns 10 ms, at the requested sample rate.
- if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
- RTC_LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
- return -1;
- }
- return 0;
-}
-
-/////////////////////////////////////////
// Statistics
//
-// TODO(turajs) change the return value to void. Also change the corresponding
-// NetEq function.
-int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
- receiver_.GetNetworkStatistics(statistics);
- return 0;
-}
-
bool AudioCodingModuleImpl::HaveValidEncoder(
absl::string_view caller_name) const {
if (!encoder_stack_) {
@@ -617,21 +525,12 @@
} // namespace
-AudioCodingModule::Config::Config(
- rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
- : neteq_config(),
- clock(Clock::GetRealTimeClock()),
- decoder_factory(decoder_factory) {
- // Post-decode VAD is disabled by default in NetEq, however, Audio
- // Conference Mixer relies on VAD decisions and fails without them.
- neteq_config.enable_post_decode_vad = true;
+std::unique_ptr<AudioCodingModule> AudioCodingModule::Create() {
+ return std::make_unique<AudioCodingModuleImpl>();
}
-AudioCodingModule::Config::Config(const Config&) = default;
-AudioCodingModule::Config::~Config() = default;
-
AudioCodingModule* AudioCodingModule::Create(const Config& config) {
- return new AudioCodingModuleImpl(config);
+ return new AudioCodingModuleImpl();
}
} // namespace webrtc
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index f1eb81c..42f4a7e 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -172,12 +172,11 @@
void TearDown() {}
void SetUp() {
- acm_.reset(AudioCodingModule::Create([this] {
- AudioCodingModule::Config config;
- config.clock = clock_;
- config.decoder_factory = CreateBuiltinAudioDecoderFactory();
- return config;
- }()));
+ acm_ = AudioCodingModule::Create();
+ acm2::AcmReceiver::Config config;
+ config.clock = *clock_;
+ config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+ acm_receiver_ = std::make_unique<acm2::AcmReceiver>(config);
rtp_utility_->Populate(&rtp_header_);
@@ -200,7 +199,7 @@
}
virtual void RegisterCodec() {
- acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}});
+ acm_receiver_->SetCodecs({{kPayloadType, *audio_format_}});
acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
kPayloadType, *audio_format_, absl::nullopt));
}
@@ -212,15 +211,16 @@
virtual void InsertPacket() {
const uint8_t kPayload[kPayloadSizeBytes] = {0};
- ASSERT_EQ(0,
- acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_));
+ ASSERT_EQ(0, acm_receiver_->InsertPacket(rtp_header_,
+ rtc::ArrayView<const uint8_t>(
+ kPayload, kPayloadSizeBytes)));
rtp_utility_->Forward(&rtp_header_);
}
virtual void PullAudio() {
AudioFrame audio_frame;
bool muted;
- ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame, &muted));
+ ASSERT_EQ(0, acm_receiver_->GetAudio(-1, &audio_frame, &muted));
ASSERT_FALSE(muted);
}
@@ -242,6 +242,7 @@
std::unique_ptr<RtpData> rtp_utility_;
std::unique_ptr<AudioCodingModule> acm_;
+ std::unique_ptr<acm2::AcmReceiver> acm_receiver_;
PacketizationCallbackStubOldApi packet_cb_;
RTPHeader rtp_header_;
AudioFrame input_frame_;
@@ -255,19 +256,6 @@
class AudioCodingModuleTestOldApiDeathTest
: public AudioCodingModuleTestOldApi {};
-TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
- AudioFrame audio_frame;
- const int kSampleRateHz = 32000;
- bool muted;
- EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
- ASSERT_FALSE(muted);
- EXPECT_EQ(0u, audio_frame.timestamp_);
- EXPECT_GT(audio_frame.num_channels_, 0u);
- EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
- audio_frame.samples_per_channel_);
- EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
-}
-
// The below test is temporarily disabled on Windows due to problems
// with clang debug builds.
// TODO(tommi): Re-enable when we've figured out what the problem is.
@@ -277,7 +265,7 @@
TEST_F(AudioCodingModuleTestOldApiDeathTest, FailOnZeroDesiredFrequency) {
AudioFrame audio_frame;
bool muted;
- RTC_EXPECT_DEATH(acm_->PlayoutData10Ms(0, &audio_frame, &muted),
+ RTC_EXPECT_DEATH(acm_receiver_->GetAudio(0, &audio_frame, &muted),
"dst_sample_rate_hz");
}
#endif
@@ -310,8 +298,8 @@
: public AudioCodingModuleTestOldApi {
protected:
void RegisterCngCodec(int rtp_payload_type) {
- acm_->SetReceiveCodecs({{kPayloadType, *audio_format_},
- {rtp_payload_type, {"cn", kSampleRateHz, 1}}});
+ acm_receiver_->SetCodecs({{kPayloadType, *audio_format_},
+ {rtp_payload_type, {"cn", kSampleRateHz, 1}}});
acm_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
AudioEncoderCngConfig config;
config.speech_encoder = std::move(*enc);
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 8b518fb..d7bee84 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -64,11 +64,11 @@
AudioCodingModule() {}
public:
+ // Deprecated. Will be deleted when downlink clients have migrated off it.
struct Config {
- explicit Config(
- rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr);
- Config(const Config&);
- ~Config();
+ Config() = default;
+ Config(const Config&) = default;
+ ~Config() = default;
NetEq::Config neteq_config;
Clock* clock;
@@ -76,13 +76,12 @@
NetEqFactory* neteq_factory = nullptr;
};
+ static std::unique_ptr<AudioCodingModule> Create();
+ // Deprecated. Will be deleted when downlink clients have migrated to the
+ // above method.
static AudioCodingModule* Create(const Config& config);
virtual ~AudioCodingModule() = default;
- ///////////////////////////////////////////////////////////////////////////
- // Sender
- //
-
// `modifier` is called exactly once with one argument: a pointer to the
// unique_ptr that holds the current encoder (which is null if there is no
// current encoder). For the duration of the call, `modifier` has exclusive
@@ -153,89 +152,9 @@
virtual int SetPacketLossRate(int packet_loss_rate) = 0;
///////////////////////////////////////////////////////////////////////////
- // Receiver
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // int32_t InitializeReceiver()
- // Any decoder-related state of ACM will be initialized to the
- // same state when ACM is created. This will not interrupt or
- // effect encoding functionality of ACM. ACM would lose all the
- // decoding-related settings by calling this function.
- // For instance, all registered codecs are deleted and have to be
- // registered again.
- //
- // Return value:
- // -1 if failed to initialize,
- // 0 if succeeded.
- //
- virtual int32_t InitializeReceiver() = 0;
-
- // Replace any existing decoders with the given payload type -> decoder map.
- virtual void SetReceiveCodecs(
- const std::map<int, SdpAudioFormat>& codecs) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // int32_t IncomingPacket()
- // Call this function to insert a parsed RTP packet into ACM.
- //
- // Inputs:
- // -incoming_payload : received payload.
- // -payload_len_bytes : the length of payload in bytes.
- // -rtp_info : the relevant information retrieved from RTP
- // header.
- //
- // Return value:
- // -1 if failed to push in the payload
- // 0 if payload is successfully pushed in.
- //
- virtual int32_t IncomingPacket(const uint8_t* incoming_payload,
- size_t payload_len_bytes,
- const RTPHeader& rtp_header) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // int32_t PlayoutData10Ms(
- // Get 10 milliseconds of raw audio data for playout, at the given sampling
- // frequency. ACM will perform a resampling if required.
- //
- // Input:
- // -desired_freq_hz : the desired sampling frequency, in Hertz, of the
- // output audio. If set to -1, the function returns
- // the audio at the current sampling frequency.
- //
- // Output:
- // -audio_frame : output audio frame which contains raw audio data
- // and other relevant parameters.
- // -muted : if true, the sample data in audio_frame is not
- // populated, and must be interpreted as all zero.
- //
- // Return value:
- // -1 if the function fails,
- // 0 if the function succeeds.
- //
- virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
- AudioFrame* audio_frame,
- bool* muted) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
// statistics
//
- ///////////////////////////////////////////////////////////////////////////
- // int32_t GetNetworkStatistics()
- // Get network statistics. Note that the internal statistics of NetEq are
- // reset by this call.
- //
- // Input:
- // -network_statistics : a structure that contains network statistics.
- //
- // Return value:
- // -1 if failed to set the network statistics,
- // 0 if statistics are set successfully.
- //
- virtual int32_t GetNetworkStatistics(
- NetworkStatistics* network_statistics) = 0;
-
virtual ANAStats GetANAStats() const = 0;
virtual int GetTargetBitrate() const = 0;
diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc
index 8adca92..e7fca1b 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -307,8 +307,7 @@
// Set up ACM.
const int timestamp_rate_hz = codec->RtpTimestampRateHz();
- AudioCodingModule::Config config;
- std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(config));
+ auto acm(AudioCodingModule::Create());
acm->SetEncoder(std::move(codec));
// Open files.
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index 35aa6cb..8f634db 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -82,8 +82,8 @@
return 0;
}
- status =
- _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtp_header);
+ status = _receiverACM->InsertPacket(
+ rtp_header, rtc::ArrayView<const uint8_t>(_payloadData, payloadDataSize));
return status;
}
@@ -228,8 +228,8 @@
Channel::~Channel() {}
-void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
- _receiverACM = acm;
+void Channel::RegisterReceiverACM(acm2::AcmReceiver* acm_receiver) {
+ _receiverACM = acm_receiver;
return;
}
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index 7a8829e..ebf4461 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -13,6 +13,7 @@
#include <stdio.h>
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/synchronization/mutex.h"
@@ -54,7 +55,7 @@
size_t payloadSize,
int64_t absolute_capture_timestamp_ms) override;
- void RegisterReceiverACM(AudioCodingModule* acm);
+ void RegisterReceiverACM(acm2::AcmReceiver* acm_receiver);
void ResetStats();
@@ -83,7 +84,7 @@
private:
void CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize);
- AudioCodingModule* _receiverACM;
+ acm2::AcmReceiver* _receiverACM;
uint16_t _seqNo;
// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
uint8_t _payloadData[60 * 32 * 2 * 2];
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 9f9c4aa..014a1d2 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -102,34 +102,32 @@
: _playoutLengthSmpls(kWebRtc10MsPcmAudio),
_payloadSizeBytes(MAX_INCOMING_PAYLOAD) {}
-void Receiver::Setup(AudioCodingModule* acm,
+void Receiver::Setup(acm2::AcmReceiver* acm_receiver,
RTPStream* rtpStream,
absl::string_view out_file_name,
size_t channels,
int file_num) {
- EXPECT_EQ(0, acm->InitializeReceiver());
-
if (channels == 1) {
- acm->SetReceiveCodecs({{107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {0, {"PCMU", 8000, 1}},
- {8, {"PCMA", 8000, 1}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {120, {"OPUS", 48000, 2}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ acm_receiver->SetCodecs({{107, {"L16", 8000, 1}},
+ {108, {"L16", 16000, 1}},
+ {109, {"L16", 32000, 1}},
+ {0, {"PCMU", 8000, 1}},
+ {8, {"PCMA", 8000, 1}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {120, {"OPUS", 48000, 2}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
} else {
ASSERT_EQ(channels, 2u);
- acm->SetReceiveCodecs({{111, {"L16", 8000, 2}},
- {112, {"L16", 16000, 2}},
- {113, {"L16", 32000, 2}},
- {110, {"PCMU", 8000, 2}},
- {118, {"PCMA", 8000, 2}},
- {119, {"G722", 8000, 2}},
- {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
+ acm_receiver->SetCodecs({{111, {"L16", 8000, 2}},
+ {112, {"L16", 16000, 2}},
+ {113, {"L16", 32000, 2}},
+ {110, {"PCMU", 8000, 2}},
+ {118, {"PCMA", 8000, 2}},
+ {119, {"G722", 8000, 2}},
+ {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
}
int playSampFreq;
@@ -146,7 +144,7 @@
_realPayloadSizeBytes = 0;
_playoutBuffer = new int16_t[kWebRtc10MsPcmAudio];
_frequency = playSampFreq;
- _acm = acm;
+ _acm_receiver = acm_receiver;
_firstTime = true;
}
@@ -171,8 +169,9 @@
}
}
- EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
- _rtpHeader));
+ EXPECT_EQ(0, _acm_receiver->InsertPacket(
+ _rtpHeader, rtc::ArrayView<const uint8_t>(
+ _incomingPayload, _realPayloadSizeBytes)));
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
@@ -185,7 +184,7 @@
bool Receiver::PlayoutData() {
AudioFrame audioFrame;
bool muted;
- int32_t ok = _acm->PlayoutData10Ms(_frequency, &audioFrame, &muted);
+ int32_t ok = _acm_receiver->GetAudio(_frequency, &audioFrame, &muted);
if (muted) {
ADD_FAILURE();
return false;
@@ -240,8 +239,7 @@
int file_num = 0;
for (const auto& send_codec : send_codecs) {
RTPFile rtpFile;
- std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+ std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
std::string fileName = webrtc::test::TempFilename(
webrtc::test::OutputPath(), "encode_decode_rtp");
@@ -256,8 +254,12 @@
rtpFile.Open(fileName.c_str(), "rb");
rtpFile.ReadHeader();
+ std::unique_ptr<acm2::AcmReceiver> acm_receiver(
+ std::make_unique<acm2::AcmReceiver>(
+ acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory())));
Receiver receiver;
- receiver.Setup(acm.get(), &rtpFile, "encodeDecode_out", 1, file_num);
+ receiver.Setup(acm_receiver.get(), &rtpFile, "encodeDecode_out", 1,
+ file_num);
receiver.Run();
receiver.Teardown();
rtpFile.Close();
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index 89b7644..9cd2c23 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -15,6 +15,7 @@
#include <string.h>
#include "absl/strings/string_view.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "modules/audio_coding/test/RTPFile.h"
@@ -73,7 +74,7 @@
public:
Receiver();
virtual ~Receiver() {}
- void Setup(AudioCodingModule* acm,
+ void Setup(acm2::AcmReceiver* acm_receiver,
RTPStream* rtpStream,
absl::string_view out_file_name,
size_t channels,
@@ -91,7 +92,7 @@
bool _firstTime;
protected:
- AudioCodingModule* _acm;
+ acm2::AcmReceiver* _acm_receiver;
uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
RTPStream* _rtpStream;
RTPHeader _rtpHeader;
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index 799e9c5..c4f6656 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -27,7 +27,7 @@
lost_packet_counter_(0),
burst_lost_counter_(burst_length_) {}
-void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
+void ReceiverWithPacketLoss::Setup(acm2::AcmReceiver* acm_receiver,
RTPStream* rtpStream,
absl::string_view out_file_name,
int channels,
@@ -39,7 +39,7 @@
burst_lost_counter_ = burst_length_; // To prevent first packet gets lost.
rtc::StringBuilder ss;
ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
- Receiver::Setup(acm, rtpStream, ss.str(), channels, file_num);
+ Receiver::Setup(acm_receiver, rtpStream, ss.str(), channels, file_num);
}
bool ReceiverWithPacketLoss::IncomingPacket() {
@@ -58,7 +58,9 @@
}
if (!PacketLost()) {
- _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpHeader);
+ _acm_receiver->InsertPacket(
+ _rtpHeader, rtc::ArrayView<const uint8_t>(_incomingPayload,
+ _realPayloadSizeBytes));
}
packet_counter_++;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
@@ -135,8 +137,7 @@
return;
#else
RTPFile rtpFile;
- std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+ std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
SdpAudioFormat send_format = SdpAudioFormat("opus", 48000, 2);
if (channels_ == 2) {
send_format.parameters = {{"stereo", "1"}};
@@ -155,8 +156,11 @@
rtpFile.Open(fileName.c_str(), "rb");
rtpFile.ReadHeader();
+ std::unique_ptr<acm2::AcmReceiver> acm_receiver(
+ std::make_unique<acm2::AcmReceiver>(
+ acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory())));
ReceiverWithPacketLoss receiver;
- receiver.Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, 15,
+ receiver.Setup(acm_receiver.get(), &rtpFile, "packetLoss_out", channels_, 15,
actual_loss_rate_, burst_length_);
receiver.Run();
receiver.Teardown();
diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
index d841d65..7569e23 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -21,7 +21,7 @@
class ReceiverWithPacketLoss : public Receiver {
public:
ReceiverWithPacketLoss();
- void Setup(AudioCodingModule* acm,
+ void Setup(acm2::AcmReceiver* acm_receiver,
RTPStream* rtpStream,
absl::string_view out_file_name,
int channels,
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index b44037d..dd51760 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -55,8 +55,8 @@
TestPack::~TestPack() {}
-void TestPack::RegisterReceiverACM(AudioCodingModule* acm) {
- receiver_acm_ = acm;
+void TestPack::RegisterReceiverACM(acm2::AcmReceiver* acm_receiver) {
+ receiver_acm_ = acm_receiver;
return;
}
@@ -83,8 +83,8 @@
// Only run mono for all test cases.
memcpy(payload_data_, payload_data, payload_size);
- status =
- receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_header);
+ status = receiver_acm_->InsertPacket(
+ rtp_header, rtc::ArrayView<const uint8_t>(payload_data_, payload_size));
payload_size_ = payload_size;
timestamp_diff_ = timestamp - last_in_timestamp_;
@@ -106,10 +106,9 @@
}
TestAllCodecs::TestAllCodecs()
- : acm_a_(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
- acm_b_(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+ : acm_a_(AudioCodingModule::Create()),
+ acm_b_(std::make_unique<acm2::AcmReceiver>(
+ acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()))),
channel_a_to_b_(NULL),
test_count_(0),
packet_size_samples_(0),
@@ -127,26 +126,23 @@
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
infile_a_.Open(file_name, 32000, "rb");
- acm_a_->InitializeReceiver();
- acm_b_->InitializeReceiver();
-
- acm_b_->SetReceiveCodecs({{107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {111, {"L16", 8000, 2}},
- {112, {"L16", 16000, 2}},
- {113, {"L16", 32000, 2}},
- {0, {"PCMU", 8000, 1}},
- {110, {"PCMU", 8000, 2}},
- {8, {"PCMA", 8000, 1}},
- {118, {"PCMA", 8000, 2}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {119, {"G722", 8000, 2}},
- {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ acm_b_->SetCodecs({{107, {"L16", 8000, 1}},
+ {108, {"L16", 16000, 1}},
+ {109, {"L16", 32000, 1}},
+ {111, {"L16", 8000, 2}},
+ {112, {"L16", 16000, 2}},
+ {113, {"L16", 32000, 2}},
+ {0, {"PCMU", 8000, 1}},
+ {110, {"PCMU", 8000, 2}},
+ {8, {"PCMA", 8000, 1}},
+ {118, {"PCMA", 8000, 2}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {119, {"G722", 8000, 2}},
+ {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
// Create and connect the channel
channel_a_to_b_ = new TestPack;
@@ -158,113 +154,113 @@
test_count_++;
OpenOutFile(test_count_);
char codec_g722[] = "G722";
- RegisterSendCodec('A', codec_g722, 16000, 64000, 160, 0);
+ RegisterSendCodec(codec_g722, 16000, 64000, 160, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_g722, 16000, 64000, 320, 0);
+ RegisterSendCodec(codec_g722, 16000, 64000, 320, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_g722, 16000, 64000, 480, 0);
+ RegisterSendCodec(codec_g722, 16000, 64000, 480, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_g722, 16000, 64000, 640, 0);
+ RegisterSendCodec(codec_g722, 16000, 64000, 640, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_g722, 16000, 64000, 800, 0);
+ RegisterSendCodec(codec_g722, 16000, 64000, 800, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0);
+ RegisterSendCodec(codec_g722, 16000, 64000, 960, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
#ifdef WEBRTC_CODEC_ILBC
test_count_++;
OpenOutFile(test_count_);
char codec_ilbc[] = "ILBC";
- RegisterSendCodec('A', codec_ilbc, 8000, 13300, 240, 0);
+ RegisterSendCodec(codec_ilbc, 8000, 13300, 240, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_ilbc, 8000, 13300, 480, 0);
+ RegisterSendCodec(codec_ilbc, 8000, 13300, 480, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_ilbc, 8000, 15200, 160, 0);
+ RegisterSendCodec(codec_ilbc, 8000, 15200, 160, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_ilbc, 8000, 15200, 320, 0);
+ RegisterSendCodec(codec_ilbc, 8000, 15200, 320, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
test_count_++;
OpenOutFile(test_count_);
char codec_l16[] = "L16";
- RegisterSendCodec('A', codec_l16, 8000, 128000, 80, 0);
+ RegisterSendCodec(codec_l16, 8000, 128000, 80, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_l16, 8000, 128000, 160, 0);
+ RegisterSendCodec(codec_l16, 8000, 128000, 160, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_l16, 8000, 128000, 240, 0);
+ RegisterSendCodec(codec_l16, 8000, 128000, 240, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_l16, 8000, 128000, 320, 0);
+ RegisterSendCodec(codec_l16, 8000, 128000, 320, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
test_count_++;
OpenOutFile(test_count_);
- RegisterSendCodec('A', codec_l16, 16000, 256000, 160, 0);
+ RegisterSendCodec(codec_l16, 16000, 256000, 160, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_l16, 16000, 256000, 320, 0);
+ RegisterSendCodec(codec_l16, 16000, 256000, 320, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_l16, 16000, 256000, 480, 0);
+ RegisterSendCodec(codec_l16, 16000, 256000, 480, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_l16, 16000, 256000, 640, 0);
+ RegisterSendCodec(codec_l16, 16000, 256000, 640, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
test_count_++;
OpenOutFile(test_count_);
- RegisterSendCodec('A', codec_l16, 32000, 512000, 320, 0);
+ RegisterSendCodec(codec_l16, 32000, 512000, 320, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0);
+ RegisterSendCodec(codec_l16, 32000, 512000, 640, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
test_count_++;
OpenOutFile(test_count_);
char codec_pcma[] = "PCMA";
- RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, 0);
+ RegisterSendCodec(codec_pcma, 8000, 64000, 80, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, 0);
+ RegisterSendCodec(codec_pcma, 8000, 64000, 160, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, 0);
+ RegisterSendCodec(codec_pcma, 8000, 64000, 240, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, 0);
+ RegisterSendCodec(codec_pcma, 8000, 64000, 320, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, 0);
+ RegisterSendCodec(codec_pcma, 8000, 64000, 400, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, 0);
+ RegisterSendCodec(codec_pcma, 8000, 64000, 480, 0);
Run(channel_a_to_b_);
char codec_pcmu[] = "PCMU";
- RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, 0);
+ RegisterSendCodec(codec_pcmu, 8000, 64000, 80, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, 0);
+ RegisterSendCodec(codec_pcmu, 8000, 64000, 160, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, 0);
+ RegisterSendCodec(codec_pcmu, 8000, 64000, 240, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, 0);
+ RegisterSendCodec(codec_pcmu, 8000, 64000, 320, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, 0);
+ RegisterSendCodec(codec_pcmu, 8000, 64000, 400, 0);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_pcmu, 8000, 64000, 480, 0);
+ RegisterSendCodec(codec_pcmu, 8000, 64000, 480, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
#ifdef WEBRTC_CODEC_OPUS
test_count_++;
OpenOutFile(test_count_);
char codec_opus[] = "OPUS";
- RegisterSendCodec('A', codec_opus, 48000, 6000, 480, kVariableSize);
+ RegisterSendCodec(codec_opus, 48000, 6000, 480, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 20000, 480 * 2, kVariableSize);
+ RegisterSendCodec(codec_opus, 48000, 20000, 480 * 2, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 32000, 480 * 4, kVariableSize);
+ RegisterSendCodec(codec_opus, 48000, 32000, 480 * 4, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 48000, 480, kVariableSize);
+ RegisterSendCodec(codec_opus, 48000, 48000, 480, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 4, kVariableSize);
+ RegisterSendCodec(codec_opus, 48000, 64000, 480 * 4, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 96000, 480 * 6, kVariableSize);
+ RegisterSendCodec(codec_opus, 48000, 96000, 480 * 6, kVariableSize);
Run(channel_a_to_b_);
- RegisterSendCodec('A', codec_opus, 48000, 500000, 480 * 2, kVariableSize);
+ RegisterSendCodec(codec_opus, 48000, 500000, 480 * 2, kVariableSize);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
@@ -272,8 +268,7 @@
// Register Codec to use in the test
//
-// Input: side - which ACM to use, 'A' or 'B'
-// codec_name - name to use when register the codec
+// Input: codec_name - name to use when register the codec
// sampling_freq_hz - sampling frequency in Herz
// rate - bitrate in bytes
// packet_size - packet size in samples
@@ -281,8 +276,7 @@
// used when registering, can be an internal header
// set to kVariableSize if the codec is a variable
// rate codec
-void TestAllCodecs::RegisterSendCodec(char side,
- char* codec_name,
+void TestAllCodecs::RegisterSendCodec(char* codec_name,
int32_t sampling_freq_hz,
int rate,
int packet_size,
@@ -316,29 +310,12 @@
packet_size_bytes_ = kVariableSize;
}
- // Set pointer to the ACM where to register the codec.
- AudioCodingModule* my_acm = NULL;
- switch (side) {
- case 'A': {
- my_acm = acm_a_.get();
- break;
- }
- case 'B': {
- my_acm = acm_b_.get();
- break;
- }
- default: {
- break;
- }
- }
- ASSERT_TRUE(my_acm != NULL);
-
auto factory = CreateBuiltinAudioEncoderFactory();
constexpr int payload_type = 17;
SdpAudioFormat format = {codec_name, clockrate_hz, num_channels};
format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
- my_acm->SetEncoder(
+ acm_a_->SetEncoder(
factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
}
@@ -381,7 +358,7 @@
// Run received side of ACM.
bool muted;
- CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame, &muted));
+ CHECK_ERROR(acm_b_->GetAudio(out_freq_hz, &audio_frame, &muted));
ASSERT_FALSE(muted);
// Write output speech to file.
diff --git a/modules/audio_coding/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h
index 0c27641..a17038a 100644
--- a/modules/audio_coding/test/TestAllCodecs.h
+++ b/modules/audio_coding/test/TestAllCodecs.h
@@ -13,6 +13,7 @@
#include <memory>
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
@@ -23,7 +24,7 @@
TestPack();
~TestPack();
- void RegisterReceiverACM(AudioCodingModule* acm);
+ void RegisterReceiverACM(acm2::AcmReceiver* acm_receiver);
int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
@@ -37,7 +38,7 @@
void reset_payload_size();
private:
- AudioCodingModule* receiver_acm_;
+ acm2::AcmReceiver* receiver_acm_;
uint16_t sequence_number_;
uint8_t payload_data_[60 * 32 * 2 * 2];
uint32_t timestamp_diff_;
@@ -58,8 +59,7 @@
// codec name, and a sampling frequency matching is not required.
// This is useful for codecs which support several sampling frequency.
// Note! Only mono mode is tested in this test.
- void RegisterSendCodec(char side,
- char* codec_name,
+ void RegisterSendCodec(char* codec_name,
int32_t sampling_freq_hz,
int rate,
int packet_size,
@@ -69,7 +69,7 @@
void OpenOutFile(int test_number);
std::unique_ptr<AudioCodingModule> acm_a_;
- std::unique_ptr<AudioCodingModule> acm_b_;
+ std::unique_ptr<acm2::AcmReceiver> acm_b_;
TestPack* channel_a_to_b_;
PCMFile infile_a_;
PCMFile outfile_b_;
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index fff48b2..f8acf48 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -42,10 +42,9 @@
AudioDecoderG722,
AudioDecoderL16,
AudioDecoderOpus>()),
- _acmA(AudioCodingModule::Create(
- AudioCodingModule::Config(decoder_factory_))),
- _acmB(AudioCodingModule::Create(
- AudioCodingModule::Config(decoder_factory_))),
+ _acmA(AudioCodingModule::Create()),
+ _acm_receiver(std::make_unique<acm2::AcmReceiver>(
+ acm2::AcmReceiver::Config(decoder_factory_))),
_channelA2B(NULL),
_testCntr(0) {}
@@ -61,13 +60,10 @@
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
_inFileA.Open(file_name, 32000, "rb");
- ASSERT_EQ(0, _acmA->InitializeReceiver());
- ASSERT_EQ(0, _acmB->InitializeReceiver());
-
// Create and connect the channel
_channelA2B = new Channel;
_acmA->RegisterTransportCallback(_channelA2B);
- _channelA2B->RegisterReceiverACM(_acmB.get());
+ _channelA2B->RegisterReceiverACM(_acm_receiver.get());
RegisterSendCodec(_acmA, {"L16", 8000, 1}, Vad::kVadAggressive, true);
@@ -136,7 +132,6 @@
absl::optional<Vad::Aggressiveness> vad_mode,
bool use_red) {
constexpr int payload_type = 17, cn_payload_type = 27, red_payload_type = 37;
- const auto& other_acm = &acm == &_acmA ? _acmB : _acmA;
auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
absl::nullopt);
@@ -165,7 +160,7 @@
}
}
acm->SetEncoder(std::move(encoder));
- other_acm->SetReceiveCodecs(receive_codecs);
+ _acm_receiver->SetCodecs(receive_codecs);
}
void TestRedFec::Run() {
@@ -180,7 +175,7 @@
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
bool muted;
- EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+ EXPECT_EQ(0, _acm_receiver->GetAudio(outFreqHzB, &audioFrame, &muted));
ASSERT_FALSE(muted);
_outFileB.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_);
}
diff --git a/modules/audio_coding/test/TestRedFec.h b/modules/audio_coding/test/TestRedFec.h
index dbadd88..173b03f 100644
--- a/modules/audio_coding/test/TestRedFec.h
+++ b/modules/audio_coding/test/TestRedFec.h
@@ -17,13 +17,14 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "common_audio/vad/include/vad.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "test/scoped_key_value_config.h"
namespace webrtc {
-class TestRedFec {
+class TestRedFec final {
public:
explicit TestRedFec();
~TestRedFec();
@@ -42,7 +43,7 @@
const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
std::unique_ptr<AudioCodingModule> _acmA;
- std::unique_ptr<AudioCodingModule> _acmB;
+ std::unique_ptr<acm2::AcmReceiver> _acm_receiver;
Channel* _channelA2B;
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 599fafb..94a1576 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -35,8 +35,8 @@
TestPackStereo::~TestPackStereo() {}
-void TestPackStereo::RegisterReceiverACM(AudioCodingModule* acm) {
- receiver_acm_ = acm;
+void TestPackStereo::RegisterReceiverACM(acm2::AcmReceiver* acm_receiver) {
+ receiver_acm_ = acm_receiver;
return;
}
@@ -60,8 +60,8 @@
}
if (lost_packet_ == false) {
- status =
- receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_header);
+ status = receiver_acm_->InsertPacket(
+ rtp_header, rtc::ArrayView<const uint8_t>(payload_data, payload_size));
if (frame_type != AudioFrameType::kAudioFrameCN) {
payload_size_ = static_cast<int>(payload_size);
@@ -97,10 +97,9 @@
}
TestStereo::TestStereo()
- : acm_a_(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
- acm_b_(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+ : acm_a_(AudioCodingModule::Create()),
+ acm_b_(std::make_unique<acm2::AcmReceiver>(
+ acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()))),
channel_a2b_(NULL),
test_cntr_(0),
pack_size_samp_(0),
@@ -134,28 +133,27 @@
// Create and initialize two ACMs, one for each side of a one-to-one call.
ASSERT_TRUE((acm_a_.get() != NULL) && (acm_b_.get() != NULL));
- EXPECT_EQ(0, acm_a_->InitializeReceiver());
- EXPECT_EQ(0, acm_b_->InitializeReceiver());
+ acm_b_->FlushBuffers();
- acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {111, {"L16", 8000, 2}},
- {112, {"L16", 16000, 2}},
- {113, {"L16", 32000, 2}},
- {0, {"PCMU", 8000, 1}},
- {110, {"PCMU", 8000, 2}},
- {8, {"PCMA", 8000, 1}},
- {118, {"PCMA", 8000, 2}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {119, {"G722", 8000, 2}},
- {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ acm_b_->SetCodecs({{103, {"ISAC", 16000, 1}},
+ {104, {"ISAC", 32000, 1}},
+ {107, {"L16", 8000, 1}},
+ {108, {"L16", 16000, 1}},
+ {109, {"L16", 32000, 1}},
+ {111, {"L16", 8000, 2}},
+ {112, {"L16", 16000, 2}},
+ {113, {"L16", 32000, 2}},
+ {0, {"PCMU", 8000, 1}},
+ {110, {"PCMU", 8000, 2}},
+ {8, {"PCMA", 8000, 1}},
+ {118, {"PCMA", 8000, 2}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {119, {"G722", 8000, 2}},
+ {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
@@ -389,7 +387,7 @@
OpenOutFile(test_cntr_);
// Encode and decode in mono.
RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels);
- acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
+ acm_b_->SetCodecs({{120, {"OPUS", 48000, 2}}});
Run(channel_a2b_, audio_channels, codec_channels);
// Encode in stereo, decode in mono.
@@ -408,13 +406,13 @@
// Decode in stereo.
test_cntr_++;
OpenOutFile(test_cntr_);
- acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
+ acm_b_->SetCodecs({{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
Run(channel_a2b_, audio_channels, 2);
out_file_.Close();
// Decode in mono.
test_cntr_++;
OpenOutFile(test_cntr_);
- acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
+ acm_b_->SetCodecs({{120, {"OPUS", 48000, 2}}});
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
@@ -455,7 +453,9 @@
break;
}
case 'B': {
- my_acm = acm_b_.get();
+ // We no longer use this case. Refactor code to avoid the switch.
+ ASSERT_TRUE(false);
+ // my_acm = acm_b_.get();
break;
}
default:
@@ -559,7 +559,7 @@
// Run receive side of ACM
bool muted;
- EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+ EXPECT_EQ(0, acm_b_->GetAudio(out_freq_hz_b, &audio_frame, &muted));
ASSERT_FALSE(muted);
// Write output speech to file
diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
index 4c50a4b..a215c90 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -15,6 +15,7 @@
#include <memory>
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
@@ -29,7 +30,7 @@
TestPackStereo();
~TestPackStereo();
- void RegisterReceiverACM(AudioCodingModule* acm);
+ void RegisterReceiverACM(acm2::AcmReceiver* acm_receiver);
int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
@@ -45,7 +46,7 @@
void set_lost_packet(bool lost);
private:
- AudioCodingModule* receiver_acm_;
+ acm2::AcmReceiver* receiver_acm_;
int16_t seq_no_;
uint32_t timestamp_diff_;
uint32_t last_in_timestamp_;
@@ -81,7 +82,7 @@
void OpenOutFile(int16_t test_number);
std::unique_ptr<AudioCodingModule> acm_a_;
- std::unique_ptr<AudioCodingModule> acm_b_;
+ std::unique_ptr<acm2::AcmReceiver> acm_b_;
TestPackStereo* channel_a2b_;
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index de26caf..1789efd 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -70,10 +70,9 @@
CreateAudioEncoderFactory<AudioEncoderIlbc, AudioEncoderOpus>()),
decoder_factory_(
CreateAudioDecoderFactory<AudioDecoderIlbc, AudioDecoderOpus>()),
- acm_send_(AudioCodingModule::Create(
- AudioCodingModule::Config(decoder_factory_))),
- acm_receive_(AudioCodingModule::Create(
- AudioCodingModule::Config(decoder_factory_))),
+ acm_send_(AudioCodingModule::Create()),
+ acm_receive_(std::make_unique<acm2::AcmReceiver>(
+ acm2::AcmReceiver::Config(decoder_factory_))),
channel_(std::make_unique<Channel>()),
packetization_callback_(
std::make_unique<MonitoringAudioPacketizationCallback>(
@@ -104,7 +103,7 @@
acm_send_->SetEncoder(std::move(encoder));
std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}};
- acm_receive_->SetReceiveCodecs(receive_codecs);
+ acm_receive_->SetCodecs(receive_codecs);
return added_comfort_noise;
}
@@ -143,7 +142,7 @@
time_stamp_ += frame_size_samples;
EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
bool muted;
- acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame, &muted);
+ acm_receive_->GetAudio(kOutputFreqHz, &audio_frame, &muted);
ASSERT_FALSE(muted);
out_file.Write10MsData(audio_frame);
}
diff --git a/modules/audio_coding/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h
index d81ae28..17b3f41 100644
--- a/modules/audio_coding/test/TestVADDTX.h
+++ b/modules/audio_coding/test/TestVADDTX.h
@@ -17,6 +17,7 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "common_audio/vad/include/vad.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/test/Channel.h"
@@ -84,7 +85,7 @@
const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
std::unique_ptr<AudioCodingModule> acm_send_;
- std::unique_ptr<AudioCodingModule> acm_receive_;
+ std::unique_ptr<acm2::AcmReceiver> acm_receive_;
std::unique_ptr<Channel> channel_;
std::unique_ptr<MonitoringAudioPacketizationCallback> packetization_callback_;
uint32_t time_stamp_ = 0x12345678;
diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc
index 7612aa4..9dbc645 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -20,7 +20,6 @@
#include "modules/audio_coding/test/TestRedFec.h"
#include "modules/audio_coding/test/TestStereo.h"
#include "modules/audio_coding/test/TestVADDTX.h"
-#include "modules/audio_coding/test/TwoWayCommunication.h"
#include "modules/audio_coding/test/opus_test.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
diff --git a/modules/audio_coding/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
deleted file mode 100644
index b42415a..0000000
--- a/modules/audio_coding/test/TwoWayCommunication.cc
+++ /dev/null
@@ -1,191 +0,0 @@
-/*
- * Copyright (c) 2012 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 "TwoWayCommunication.h"
-
-#include <stdio.h>
-#include <string.h>
-
-#include <memory>
-
-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "modules/audio_coding/test/PCMFile.h"
-#include "test/gtest.h"
-#include "test/testsupport/file_utils.h"
-
-namespace webrtc {
-
-#define MAX_FILE_NAME_LENGTH_BYTE 500
-
-TwoWayCommunication::TwoWayCommunication()
- : _acmA(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
- _acmRefA(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {
- AudioCodingModule::Config config;
- // The clicks will be more obvious if time-stretching is not allowed.
- // TODO(henrik.lundin) Really?
- config.neteq_config.for_test_no_time_stretching = true;
- config.decoder_factory = CreateBuiltinAudioDecoderFactory();
- _acmB.reset(AudioCodingModule::Create(config));
- _acmRefB.reset(AudioCodingModule::Create(config));
-}
-
-TwoWayCommunication::~TwoWayCommunication() {
- delete _channel_A2B;
- delete _channel_B2A;
- delete _channelRef_A2B;
- delete _channelRef_B2A;
- _inFileA.Close();
- _inFileB.Close();
- _outFileA.Close();
- _outFileB.Close();
- _outFileRefA.Close();
- _outFileRefB.Close();
-}
-
-void TwoWayCommunication::SetUpAutotest(
- AudioEncoderFactory* const encoder_factory,
- const SdpAudioFormat& format1,
- const int payload_type1,
- const SdpAudioFormat& format2,
- const int payload_type2) {
- //--- Set A codecs
- _acmA->SetEncoder(
- encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
- _acmA->SetReceiveCodecs({{payload_type2, format2}});
-
- //--- Set ref-A codecs
- _acmRefA->SetEncoder(
- encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
- _acmRefA->SetReceiveCodecs({{payload_type2, format2}});
-
- //--- Set B codecs
- _acmB->SetEncoder(
- encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
- _acmB->SetReceiveCodecs({{payload_type1, format1}});
-
- //--- Set ref-B codecs
- _acmRefB->SetEncoder(
- encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
- _acmRefB->SetReceiveCodecs({{payload_type1, format1}});
-
- uint16_t frequencyHz;
-
- //--- Input A and B
- std::string in_file_name =
- webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
- frequencyHz = 16000;
- _inFileA.Open(in_file_name, frequencyHz, "rb");
- _inFileB.Open(in_file_name, frequencyHz, "rb");
-
- //--- Output A
- std::string output_file_a = webrtc::test::OutputPath() + "outAutotestA.pcm";
- frequencyHz = 16000;
- _outFileA.Open(output_file_a, frequencyHz, "wb");
- std::string output_ref_file_a =
- webrtc::test::OutputPath() + "ref_outAutotestA.pcm";
- _outFileRefA.Open(output_ref_file_a, frequencyHz, "wb");
-
- //--- Output B
- std::string output_file_b = webrtc::test::OutputPath() + "outAutotestB.pcm";
- frequencyHz = 16000;
- _outFileB.Open(output_file_b, frequencyHz, "wb");
- std::string output_ref_file_b =
- webrtc::test::OutputPath() + "ref_outAutotestB.pcm";
- _outFileRefB.Open(output_ref_file_b, frequencyHz, "wb");
-
- //--- Set A-to-B channel
- _channel_A2B = new Channel;
- _acmA->RegisterTransportCallback(_channel_A2B);
- _channel_A2B->RegisterReceiverACM(_acmB.get());
- //--- Do the same for the reference
- _channelRef_A2B = new Channel;
- _acmRefA->RegisterTransportCallback(_channelRef_A2B);
- _channelRef_A2B->RegisterReceiverACM(_acmRefB.get());
-
- //--- Set B-to-A channel
- _channel_B2A = new Channel;
- _acmB->RegisterTransportCallback(_channel_B2A);
- _channel_B2A->RegisterReceiverACM(_acmA.get());
- //--- Do the same for reference
- _channelRef_B2A = new Channel;
- _acmRefB->RegisterTransportCallback(_channelRef_B2A);
- _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
-}
-
-void TwoWayCommunication::Perform() {
- const SdpAudioFormat format1("ISAC", 16000, 1);
- const SdpAudioFormat format2("L16", 8000, 1);
- constexpr int payload_type1 = 17, payload_type2 = 18;
-
- auto encoder_factory = CreateBuiltinAudioEncoderFactory();
-
- SetUpAutotest(encoder_factory.get(), format1, payload_type1, format2,
- payload_type2);
-
- unsigned int msecPassed = 0;
- unsigned int secPassed = 0;
-
- int32_t outFreqHzA = _outFileA.SamplingFrequency();
- int32_t outFreqHzB = _outFileB.SamplingFrequency();
-
- AudioFrame audioFrame;
-
- // In the following loop we tests that the code can handle misuse of the APIs.
- // In the middle of a session with data flowing between two sides, called A
- // and B, APIs will be called, and the code should continue to run, and be
- // able to recover.
- while (!_inFileA.EndOfFile() && !_inFileB.EndOfFile()) {
- msecPassed += 10;
- EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
- EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
- EXPECT_GE(_acmRefA->Add10MsData(audioFrame), 0);
-
- EXPECT_GT(_inFileB.Read10MsData(audioFrame), 0);
-
- EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
- EXPECT_GE(_acmRefB->Add10MsData(audioFrame), 0);
- bool muted;
- EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
- ASSERT_FALSE(muted);
- _outFileA.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
- ASSERT_FALSE(muted);
- _outFileRefA.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
- ASSERT_FALSE(muted);
- _outFileB.Write10MsData(audioFrame);
- EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
- ASSERT_FALSE(muted);
- _outFileRefB.Write10MsData(audioFrame);
-
- // Update time counters each time a second of data has passed.
- if (msecPassed >= 1000) {
- msecPassed = 0;
- secPassed++;
- }
- // Re-register send codec on side B.
- if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
- _acmB->SetEncoder(encoder_factory->MakeAudioEncoder(
- payload_type2, format2, absl::nullopt));
- }
- // Initialize receiver on side A.
- if (((secPassed % 7) == 6) && (msecPassed == 0))
- EXPECT_EQ(0, _acmA->InitializeReceiver());
- // Re-register codec on side A.
- if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
- _acmA->SetReceiveCodecs({{payload_type2, format2}});
- }
- }
-}
-
-} // namespace webrtc
diff --git a/modules/audio_coding/test/TwoWayCommunication.h b/modules/audio_coding/test/TwoWayCommunication.h
deleted file mode 100644
index b7eb9e5..0000000
--- a/modules/audio_coding/test/TwoWayCommunication.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 2012 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.
- */
-
-#ifndef MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
-#define MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
-
-#include <memory>
-
-#include "api/audio_codecs/audio_encoder_factory.h"
-#include "api/audio_codecs/audio_format.h"
-#include "modules/audio_coding/include/audio_coding_module.h"
-#include "modules/audio_coding/test/Channel.h"
-#include "modules/audio_coding/test/PCMFile.h"
-
-namespace webrtc {
-
-class TwoWayCommunication {
- public:
- TwoWayCommunication();
- ~TwoWayCommunication();
-
- void Perform();
-
- private:
- void SetUpAutotest(AudioEncoderFactory* const encoder_factory,
- const SdpAudioFormat& format1,
- int payload_type1,
- const SdpAudioFormat& format2,
- int payload_type2);
-
- std::unique_ptr<AudioCodingModule> _acmA;
- std::unique_ptr<AudioCodingModule> _acmB;
-
- std::unique_ptr<AudioCodingModule> _acmRefA;
- std::unique_ptr<AudioCodingModule> _acmRefB;
-
- Channel* _channel_A2B;
- Channel* _channel_B2A;
-
- Channel* _channelRef_A2B;
- Channel* _channelRef_B2A;
-
- PCMFile _inFileA;
- PCMFile _inFileB;
-
- PCMFile _outFileA;
- PCMFile _outFileB;
-
- PCMFile _outFileRefA;
- PCMFile _outFileRefB;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 6822bc3..dfebb5f 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -22,8 +22,8 @@
namespace webrtc {
OpusTest::OpusTest()
- : acm_receiver_(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+ : acm_receiver_(std::make_unique<acm2::AcmReceiver>(
+ acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()))),
channel_a2b_(NULL),
counter_(0),
payload_type_(255),
@@ -83,13 +83,13 @@
WebRtcOpus_DecoderInit(opus_stereo_decoder_);
ASSERT_TRUE(acm_receiver_.get() != NULL);
- EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
+ acm_receiver_->FlushBuffers();
// Register Opus stereo as receiving codec.
constexpr int kOpusPayloadType = 120;
const SdpAudioFormat kOpusFormatStereo("opus", 48000, 2, {{"stereo", "1"}});
payload_type_ = kOpusPayloadType;
- acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatStereo}});
+ acm_receiver_->SetCodecs({{kOpusPayloadType, kOpusFormatStereo}});
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
@@ -154,7 +154,7 @@
// Register Opus mono as receiving codec.
const SdpAudioFormat kOpusFormatMono("opus", 48000, 2);
- acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatMono}});
+ acm_receiver_->SetCodecs({{kOpusPayloadType, kOpusFormatMono}});
// Run Opus with 2.5 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 120);
@@ -353,8 +353,7 @@
// Run received side of ACM.
bool muted;
- ASSERT_EQ(
- 0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+ ASSERT_EQ(0, acm_receiver_->GetAudio(out_freq_hz_b, &audio_frame, &muted));
ASSERT_FALSE(muted);
// Write output speech to file.
diff --git a/modules/audio_coding/test/opus_test.h b/modules/audio_coding/test/opus_test.h
index c69f922..cf5581a 100644
--- a/modules/audio_coding/test/opus_test.h
+++ b/modules/audio_coding/test/opus_test.h
@@ -15,6 +15,7 @@
#include <memory>
+#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/test/PCMFile.h"
@@ -38,7 +39,7 @@
void OpenOutFile(int test_number);
- std::unique_ptr<AudioCodingModule> acm_receiver_;
+ std::unique_ptr<acm2::AcmReceiver> acm_receiver_;
TestPackStereo* channel_a2b_;
PCMFile in_file_stereo_;
PCMFile in_file_mono_;
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 5eccdcf..2a71627 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -25,7 +25,7 @@
protected:
TargetDelayTest()
: receiver_(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())) {}
+ acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory())) {}
~TargetDelayTest() {}
