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Adding a config struct to NetEq

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With this change, the parameters sent to the NetEq::Create method are
collected in one NetEq::Config struct. The benefit is that it is easier
to set, change and override default values, and easier to expand with
more parameters in the future.

BUG=3083
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5902 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2014-04-14 18:49:17 +00:00
parent 810acbc93e
commit 35ead381f8
11 changed files with 52 additions and 27 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
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@@ -35,7 +35,6 @@ namespace acm2 {
namespace {
const int kNeteqInitSampleRateHz = 16000;
const int kNackThresholdPackets = 2;
// |vad_activity_| field of |audio_frame| is set to |previous_audio_activity_|
@@ -119,13 +118,14 @@ bool IsCng(int codec_id) {
AcmReceiver::AcmReceiver()
: id_(0),
neteq_(NetEq::Create(kNeteqInitSampleRateHz)),
neteq_config_(),
neteq_(NetEq::Create(neteq_config_)),
last_audio_decoder_(-1), // Invalid value.
decode_lock_(RWLockWrapper::CreateRWLock()),
neteq_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
vad_enabled_(true),
previous_audio_activity_(AudioFrame::kVadPassive),
current_sample_rate_hz_(kNeteqInitSampleRateHz),
current_sample_rate_hz_(neteq_config_.sample_rate_hz),
nack_(),
nack_enabled_(false),
av_sync_(false),
@@ -774,7 +774,7 @@ bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) {
current_sample_rate_hz_ = ACMCodecDB::database_[last_audio_decoder_].plfreq;
frame->num_channels_ = decoders_[last_audio_decoder_].channels;
} else {
current_sample_rate_hz_ = kNeteqInitSampleRateHz;
current_sample_rate_hz_ = neteq_config_.sample_rate_hz;
frame->num_channels_ = 1;
}

1
webrtc/modules/audio_coding/main/acm2/acm_receiver.h
View File

@@ -340,6 +340,7 @@ class AcmReceiver {
void InsertStreamOfSyncPackets(InitialDelayManager::SyncStream* sync_stream);
int id_;
NetEq::Config neteq_config_;
NetEq* neteq_;
Decoder decoders_[ACMCodecDB::kMaxNumCodecs];
int last_audio_decoder_;

18
webrtc/modules/audio_coding/neteq4/interface/neteq.h
View File

@@ -67,6 +67,15 @@ enum NetEqBackgroundNoiseMode {
// This is the interface class for NetEq.
class NetEq {
public:
struct Config {
Config()
: sample_rate_hz(16000),
enable_audio_classifier(false) {}
int sample_rate_hz; // Initial vale. Will change with input data.
bool enable_audio_classifier;
};
enum ReturnCodes {
kOK = 0,
kFail = -1,
@@ -105,11 +114,10 @@ class NetEq {
static const int kMaxNumPacketsInBuffer = 50; // TODO(hlundin): Remove.
static const int kMaxBytesInBuffer = 113280; // TODO(hlundin): Remove.
// Creates a new NetEq object, starting at the sample rate |sample_rate_hz|.
// (Note that it will still change the sample rate depending on what payloads
// are being inserted; |sample_rate_hz| is just for startup configuration.)
static NetEq* Create(int sample_rate_hz,
bool enable_audio_classifier = false);
// Creates a new NetEq object, with parameters set in |config|. The |config|
// object will only have to be valid for the duration of the call to this
// method.
static NetEq* Create(const NetEq::Config& config);
virtual ~NetEq() {}

6
webrtc/modules/audio_coding/neteq4/neteq.cc
View File

@@ -28,13 +28,13 @@ namespace webrtc {
// Creates all classes needed and inject them into a new NetEqImpl object.
// Return the new object.
NetEq* NetEq::Create(int sample_rate_hz, bool enable_audio_classifier) {
NetEq* NetEq::Create(const NetEq::Config& config) {
BufferLevelFilter* buffer_level_filter = new BufferLevelFilter;
DecoderDatabase* decoder_database = new DecoderDatabase;
DelayPeakDetector* delay_peak_detector = new DelayPeakDetector;
DelayManager* delay_manager = new DelayManager(kMaxNumPacketsInBuffer,
delay_peak_detector);
DtmfBuffer* dtmf_buffer = new DtmfBuffer(sample_rate_hz);
DtmfBuffer* dtmf_buffer = new DtmfBuffer(config.sample_rate_hz);
DtmfToneGenerator* dtmf_tone_generator = new DtmfToneGenerator;
PacketBuffer* packet_buffer = new PacketBuffer(kMaxNumPacketsInBuffer,
kMaxBytesInBuffer);
@@ -44,7 +44,7 @@ NetEq* NetEq::Create(int sample_rate_hz, bool enable_audio_classifier) {
ExpandFactory* expand_factory = new ExpandFactory;
PreemptiveExpandFactory* preemptive_expand_factory =
new PreemptiveExpandFactory;
return new NetEqImpl(sample_rate_hz,
return new NetEqImpl(config.sample_rate_hz,
buffer_level_filter,
decoder_database,
delay_manager,

6
webrtc/modules/audio_coding/neteq4/neteq_external_decoder_unittest.cc
View File

@@ -45,13 +45,15 @@ class NetEqExternalDecoderTest : public ::testing::Test {
frame_size_ms_(10),
frame_size_samples_(frame_size_ms_ * samples_per_ms_),
output_size_samples_(frame_size_ms_ * samples_per_ms_),
neteq_external_(NetEq::Create(sample_rate_hz_)),
neteq_(NetEq::Create(sample_rate_hz_)),
external_decoder_(new MockExternalPcm16B(kDecoderPCM16Bswb32kHz)),
rtp_generator_(samples_per_ms_),
payload_size_bytes_(0),
last_send_time_(0),
last_arrival_time_(0) {
NetEq::Config config;
config.sample_rate_hz = sample_rate_hz_;
neteq_external_ = NetEq::Create(config);
neteq_ = NetEq::Create(config);
input_ = new int16_t[frame_size_samples_];
encoded_ = new uint8_t[2 * frame_size_samples_];
}

16
webrtc/modules/audio_coding/neteq4/neteq_impl_unittest.cc
View File

@@ -48,9 +48,9 @@ int DeletePacketsAndReturnOk(PacketList* packet_list) {
class NetEqImplTest : public ::testing::Test {
protected:
static const int kInitSampleRateHz = 8000;
NetEqImplTest()
: neteq_(NULL),
config_(),
mock_buffer_level_filter_(NULL),
buffer_level_filter_(NULL),
use_mock_buffer_level_filter_(true),
@@ -75,7 +75,9 @@ class NetEqImplTest : public ::testing::Test {
mock_payload_splitter_(NULL),
payload_splitter_(NULL),
use_mock_payload_splitter_(true),
timestamp_scaler_(NULL) {}
timestamp_scaler_(NULL) {
config_.sample_rate_hz = 8000;
}
void CreateInstance() {
if (use_mock_buffer_level_filter_) {
@@ -109,10 +111,10 @@ class NetEqImplTest : public ::testing::Test {
new DelayManager(NetEq::kMaxNumPacketsInBuffer, delay_peak_detector_);
}
if (use_mock_dtmf_buffer_) {
mock_dtmf_buffer_ = new MockDtmfBuffer(kInitSampleRateHz);
mock_dtmf_buffer_ = new MockDtmfBuffer(config_.sample_rate_hz);
dtmf_buffer_ = mock_dtmf_buffer_;
} else {
dtmf_buffer_ = new DtmfBuffer(kInitSampleRateHz);
dtmf_buffer_ = new DtmfBuffer(config_.sample_rate_hz);
}
if (use_mock_dtmf_tone_generator_) {
mock_dtmf_tone_generator_ = new MockDtmfToneGenerator;
@@ -140,7 +142,7 @@ class NetEqImplTest : public ::testing::Test {
PreemptiveExpandFactory* preemptive_expand_factory =
new PreemptiveExpandFactory;
neteq_ = new NetEqImpl(kInitSampleRateHz,
neteq_ = new NetEqImpl(config_.sample_rate_hz,
buffer_level_filter_,
decoder_database_,
delay_manager_,
@@ -194,6 +196,7 @@ class NetEqImplTest : public ::testing::Test {
}
NetEqImpl* neteq_;
NetEq::Config config_;
MockBufferLevelFilter* mock_buffer_level_filter_;
BufferLevelFilter* buffer_level_filter_;
bool use_mock_buffer_level_filter_;
@@ -225,7 +228,8 @@ class NetEqImplTest : public ::testing::Test {
// This tests the interface class NetEq.
// TODO(hlundin): Move to separate file?
TEST(NetEq, CreateAndDestroy) {
NetEq* neteq = NetEq::Create(8000);
NetEq::Config config;
NetEq* neteq = NetEq::Create(config);
delete neteq;
}

6
webrtc/modules/audio_coding/neteq4/neteq_stereo_unittest.cc
View File

@@ -54,14 +54,16 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
frame_size_ms_(GetParam().frame_size),
frame_size_samples_(frame_size_ms_ * samples_per_ms_),
output_size_samples_(10 * samples_per_ms_),
neteq_mono_(NetEq::Create(sample_rate_hz_)),
neteq_(NetEq::Create(sample_rate_hz_)),
rtp_generator_mono_(samples_per_ms_),
rtp_generator_(samples_per_ms_),
payload_size_bytes_(0),
multi_payload_size_bytes_(0),
last_send_time_(0),
last_arrival_time_(0) {
NetEq::Config config;
config.sample_rate_hz = sample_rate_hz_;
neteq_mono_ = NetEq::Create(config);
neteq_ = NetEq::Create(config);
input_ = new int16_t[frame_size_samples_];
encoded_ = new uint8_t[2 * frame_size_samples_];
input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];

4
webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
View File

@@ -254,7 +254,9 @@ NetEqDecodingTest::NetEqDecodingTest()
}
void NetEqDecodingTest::SetUp() {
neteq_ = NetEq::Create(kInitSampleRateHz);
NetEq::Config config;
config.sample_rate_hz = kInitSampleRateHz;
neteq_ = NetEq::Create(config);
NetEqNetworkStatistics stat;
ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
algorithmic_delay_ms_ = stat.current_buffer_size_ms;

4
webrtc/modules/audio_coding/neteq4/tools/neteq_performance_test.cc
View File

@@ -36,7 +36,9 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
const int kPayloadType = 95;
// Initialize NetEq instance.
NetEq* neteq = NetEq::Create(kSampRateHz);
NetEq::Config config;
config.sample_rate_hz = kSampRateHz;
NetEq* neteq = NetEq::Create(config);
// Register decoder in |neteq|.
if (neteq->RegisterPayloadType(kDecoderType, kPayloadType) != 0)
return -1;

6
webrtc/modules/audio_coding/neteq4/tools/neteq_quality_test.cc
View File

@@ -42,8 +42,10 @@ NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
in_file_(new InputAudioFile(in_filename_)),
out_file_(NULL),
rtp_generator_(new RtpGenerator(in_sampling_khz_, 0, 0,
decodable_time_ms_)),
neteq_(NetEq::Create(out_sampling_khz_ * 1000)) {
decodable_time_ms_)) {
NetEq::Config config;
config.sample_rate_hz = out_sampling_khz_ * 1000;
neteq_.reset(NetEq::Create(config));
max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
in_data_.reset(new int16_t[in_size_samples_ * channels_]);
payload_.reset(new uint8_t[max_payload_bytes_]);

4
webrtc/modules/audio_coding/neteq4/tools/neteq_rtpplay.cc
View File

@@ -176,7 +176,9 @@ int main(int argc, char* argv[]) {
// Initialize NetEq instance.
int sample_rate_hz = 16000;
NetEq* neteq = NetEq::Create(sample_rate_hz);
NetEq::Config config;
config.sample_rate_hz = sample_rate_hz;
NetEq* neteq = NetEq::Create(config);
RegisterPayloadTypes(neteq);
// Read first packet.