Remove various IDs:

- AudioFrame
- AudioCodingModule

BUG=webrtc:4690
TBR=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3019543002
Cr-Commit-Position: refs/heads/master@{#20005}
This commit is contained in:
solenberg
2017-09-27 10:33:57 -07:00
committed by Commit Bot
parent 94286cb25c
commit 2d0f77585d
25 changed files with 52 additions and 90 deletions

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@ -48,8 +48,8 @@ void APITest::Wait(uint32_t waitLengthMs) {
}
APITest::APITest()
: _acmA(AudioCodingModule::Create(1)),
_acmB(AudioCodingModule::Create(2)),
: _acmA(AudioCodingModule::Create()),
_acmB(AudioCodingModule::Create()),
_channel_A2B(NULL),
_channel_B2A(NULL),
_writeToFile(true),

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@ -281,7 +281,7 @@ void EncodeDecodeTest::Perform() {
codePars[1] = 0;
codePars[2] = 0;
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
struct CodecInst sendCodecTmp;
numCodecs = acm->NumberOfCodecs();
@ -337,7 +337,7 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
int codeId,
int* codePars,
int testMode) {
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
RTPFile rtpFile;
std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
"encode_decode_rtp");

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@ -127,7 +127,7 @@ void PacketLossTest::Perform() {
#ifndef WEBRTC_CODEC_OPUS
return;
#else
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
int codec_id = acm->Codec("opus", 48000, channels_);

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@ -104,8 +104,8 @@ void TestPack::reset_payload_size() {
}
TestAllCodecs::TestAllCodecs(int test_mode)
: acm_a_(AudioCodingModule::Create(0)),
acm_b_(AudioCodingModule::Create(1)),
: acm_a_(AudioCodingModule::Create()),
acm_b_(AudioCodingModule::Create()),
channel_a_to_b_(NULL),
test_count_(0),
packet_size_samples_(0),

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@ -48,8 +48,8 @@ namespace {
}
TestRedFec::TestRedFec()
: _acmA(AudioCodingModule::Create(0)),
_acmB(AudioCodingModule::Create(1)),
: _acmA(AudioCodingModule::Create()),
_acmB(AudioCodingModule::Create()),
_channelA2B(NULL),
_testCntr(0) {
}

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@ -108,8 +108,8 @@ void TestPackStereo::set_lost_packet(bool lost) {
}
TestStereo::TestStereo(int test_mode)
: acm_a_(AudioCodingModule::Create(0)),
acm_b_(AudioCodingModule::Create(1)),
: acm_a_(AudioCodingModule::Create()),
acm_b_(AudioCodingModule::Create()),
channel_a2b_(NULL),
test_cntr_(0),
pack_size_samp_(0),

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@ -62,8 +62,8 @@ void ActivityMonitor::GetStatistics(uint32_t* counter) {
}
TestVadDtx::TestVadDtx()
: acm_send_(AudioCodingModule::Create(0)),
acm_receive_(AudioCodingModule::Create(1)),
: acm_send_(AudioCodingModule::Create()),
acm_receive_(AudioCodingModule::Create()),
channel_(new Channel),
monitor_(new ActivityMonitor) {
EXPECT_EQ(0, acm_send_->RegisterTransportCallback(channel_.get()));

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@ -34,16 +34,14 @@ namespace webrtc {
#define MAX_FILE_NAME_LENGTH_BYTE 500
TwoWayCommunication::TwoWayCommunication(int testMode)
: _acmA(AudioCodingModule::Create(1)),
_acmRefA(AudioCodingModule::Create(3)),
: _acmA(AudioCodingModule::Create()),
_acmRefA(AudioCodingModule::Create()),
_testMode(testMode) {
AudioCodingModule::Config config;
// The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
config.neteq_config.playout_mode = kPlayoutFax;
config.id = 2;
config.decoder_factory = CreateBuiltinAudioDecoderFactory();
_acmB.reset(AudioCodingModule::Create(config));
config.id = 4;
_acmRefB.reset(AudioCodingModule::Create(config));
}
@ -62,7 +60,7 @@ TwoWayCommunication::~TwoWayCommunication() {
void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
uint8_t* codecID_B) {
std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create());
uint8_t noCodec = tmpACM->NumberOfCodecs();
CodecInst codecInst;
printf("List of Supported Codecs\n");

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@ -64,8 +64,8 @@ struct TestSettings {
class DelayTest {
public:
DelayTest()
: acm_a_(AudioCodingModule::Create(0)),
acm_b_(AudioCodingModule::Create(1)),
: acm_a_(AudioCodingModule::Create()),
acm_b_(AudioCodingModule::Create()),
channel_a2b_(new Channel),
test_cntr_(0),
encoding_sample_rate_hz_(8000) {}

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@ -67,8 +67,8 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
}
ISACTest::ISACTest(int testMode)
: _acmA(AudioCodingModule::Create(1)),
_acmB(AudioCodingModule::Create(2)),
: _acmA(AudioCodingModule::Create()),
_acmB(AudioCodingModule::Create()),
_testMode(testMode) {}
ISACTest::~ISACTest() {}

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@ -61,8 +61,8 @@ class InsertPacketWithTiming {
InsertPacketWithTiming()
: sender_clock_(new SimulatedClock(0)),
receiver_clock_(new SimulatedClock(0)),
send_acm_(AudioCodingModule::Create(0, sender_clock_)),
receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
send_acm_(AudioCodingModule::Create(sender_clock_)),
receive_acm_(AudioCodingModule::Create(receiver_clock_)),
channel_(new Channel),
seq_num_fid_(fopen(FLAG_seq_num, "rt")),
send_ts_fid_(fopen(FLAG_send_ts, "rt")),

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@ -27,7 +27,7 @@
namespace webrtc {
OpusTest::OpusTest()
: acm_receiver_(AudioCodingModule::Create(0)),
: acm_receiver_(AudioCodingModule::Create()),
channel_a2b_(NULL),
counter_(0),
payload_type_(255),

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@ -22,7 +22,7 @@ namespace webrtc {
class TargetDelayTest : public ::testing::Test {
protected:
TargetDelayTest() : acm_(AudioCodingModule::Create(0)) {}
TargetDelayTest() : acm_(AudioCodingModule::Create()) {}
~TargetDelayTest() {}