Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ )
Reason for revert:
Breaks downstream
Original issue's description:
> 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}
> Committed: https://webrtc.googlesource.com/src/+/2d0f77585d556d8b11d6269d35149ae9ca14c472
TBR=henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/3014683002
Cr-Commit-Position: refs/heads/master@{#20008}
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 082506a..6cfe464 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -110,6 +110,7 @@
Clock* clock,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
AudioCodingModule::Config config;
+ config.id = 0;
config.clock = clock;
config.decoder_factory = std::move(decoder_factory);
return config;
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index 307c906..d5f196b 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -28,7 +28,7 @@
int source_rate_hz,
int test_duration_ms)
: clock_(0),
- acm_(webrtc::AudioCodingModule::Create(&clock_)),
+ acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
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 c48fbef..5997d12 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -269,6 +269,7 @@
rtc::CriticalSection acm_crit_sect_;
rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_crit_sect_);
+ int id_; // TODO(henrik.lundin) Make const.
uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_crit_sect_);
uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_crit_sect_);
acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_crit_sect_);
@@ -455,7 +456,8 @@
AudioCodingModuleImpl::AudioCodingModuleImpl(
const AudioCodingModule::Config& config)
- : expected_codec_ts_(0xD87F3F9F),
+ : id_(config.id),
+ expected_codec_ts_(0xD87F3F9F),
expected_in_ts_(0xD87F3F9F),
receiver_(config),
bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
@@ -1118,6 +1120,7 @@
LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
return -1;
}
+ audio_frame->id_ = id_;
return 0;
}
@@ -1283,7 +1286,7 @@
} // namespace
AudioCodingModule::Config::Config()
- : neteq_config(), clock(Clock::GetRealTimeClock()) {
+ : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {
// 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;
@@ -1293,15 +1296,17 @@
AudioCodingModule::Config::~Config() = default;
// Create module
-AudioCodingModule* AudioCodingModule::Create() {
+AudioCodingModule* AudioCodingModule::Create(int id) {
Config config;
+ config.id = id;
config.clock = Clock::GetRealTimeClock();
config.decoder_factory = CreateBuiltinAudioDecoderFactory();
return Create(config);
}
-AudioCodingModule* AudioCodingModule::Create(Clock* clock) {
+AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
Config config;
+ config.id = id;
config.clock = clock;
config.decoder_factory = CreateBuiltinAudioDecoderFactory();
return Create(config);
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index a010619..80fc4d8 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -157,7 +157,8 @@
class AudioCodingModuleTestOldApi : public ::testing::Test {
protected:
AudioCodingModuleTestOldApi()
- : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
+ : id_(1),
+ rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
clock_(Clock::GetRealTimeClock()) {}
~AudioCodingModuleTestOldApi() {}
@@ -165,7 +166,7 @@
void TearDown() {}
void SetUp() {
- acm_.reset(AudioCodingModule::Create(clock_));
+ acm_.reset(AudioCodingModule::Create(id_, clock_));
rtp_utility_->Populate(&rtp_header_);
@@ -229,6 +230,7 @@
VerifyEncoding();
}
+ const int id_;
std::unique_ptr<RtpUtility> rtp_utility_;
std::unique_ptr<AudioCodingModule> acm_;
PacketizationCallbackStubOldApi packet_cb_;
@@ -312,6 +314,7 @@
bool muted;
EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
ASSERT_FALSE(muted);
+ EXPECT_EQ(id_, audio_frame.id_);
EXPECT_EQ(0u, audio_frame.timestamp_);
EXPECT_GT(audio_frame.num_channels_, 0u);
EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 63af3ab..944ad60 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -70,6 +70,7 @@
Config(const Config&);
~Config();
+ int id;
NetEq::Config neteq_config;
Clock* clock;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
@@ -82,8 +83,8 @@
// injected into ACM. ACM will take the ownership of the object clock and
// delete it when destroyed.
//
- static AudioCodingModule* Create();
- static AudioCodingModule* Create(Clock* clock);
+ static AudioCodingModule* Create(int id);
+ static AudioCodingModule* Create(int id, Clock* clock);
static AudioCodingModule* Create(const Config& config);
virtual ~AudioCodingModule() = default;
diff --git a/modules/audio_coding/test/APITest.cc b/modules/audio_coding/test/APITest.cc
index b29e84e..5418342 100644
--- a/modules/audio_coding/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -48,8 +48,8 @@
}
APITest::APITest()
- : _acmA(AudioCodingModule::Create()),
- _acmB(AudioCodingModule::Create()),
+ : _acmA(AudioCodingModule::Create(1)),
+ _acmB(AudioCodingModule::Create(2)),
_channel_A2B(NULL),
_channel_B2A(NULL),
_writeToFile(true),
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 2b6b4ac..8257ed9 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -281,7 +281,7 @@
codePars[1] = 0;
codePars[2] = 0;
- std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
+ std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
struct CodecInst sendCodecTmp;
numCodecs = acm->NumberOfCodecs();
@@ -337,7 +337,7 @@
int codeId,
int* codePars,
int testMode) {
- std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
+ std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
RTPFile rtpFile;
std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
"encode_decode_rtp");
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index a6c56fa7..c80615a 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -127,7 +127,7 @@
#ifndef WEBRTC_CODEC_OPUS
return;
#else
- std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
+ std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
int codec_id = acm->Codec("opus", 48000, channels_);
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index ff28a28..74319c2 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -104,8 +104,8 @@
}
TestAllCodecs::TestAllCodecs(int test_mode)
- : acm_a_(AudioCodingModule::Create()),
- acm_b_(AudioCodingModule::Create()),
+ : acm_a_(AudioCodingModule::Create(0)),
+ acm_b_(AudioCodingModule::Create(1)),
channel_a_to_b_(NULL),
test_count_(0),
packet_size_samples_(0),
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index 58561c6..3e88290 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -48,8 +48,8 @@
}
TestRedFec::TestRedFec()
- : _acmA(AudioCodingModule::Create()),
- _acmB(AudioCodingModule::Create()),
+ : _acmA(AudioCodingModule::Create(0)),
+ _acmB(AudioCodingModule::Create(1)),
_channelA2B(NULL),
_testCntr(0) {
}
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index eca81f8..d598191 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -108,8 +108,8 @@
}
TestStereo::TestStereo(int test_mode)
- : acm_a_(AudioCodingModule::Create()),
- acm_b_(AudioCodingModule::Create()),
+ : acm_a_(AudioCodingModule::Create(0)),
+ acm_b_(AudioCodingModule::Create(1)),
channel_a2b_(NULL),
test_cntr_(0),
pack_size_samp_(0),
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index 628582d..1aa00b5 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -62,8 +62,8 @@
}
TestVadDtx::TestVadDtx()
- : acm_send_(AudioCodingModule::Create()),
- acm_receive_(AudioCodingModule::Create()),
+ : acm_send_(AudioCodingModule::Create(0)),
+ acm_receive_(AudioCodingModule::Create(1)),
channel_(new Channel),
monitor_(new ActivityMonitor) {
EXPECT_EQ(0, acm_send_->RegisterTransportCallback(channel_.get()));
diff --git a/modules/audio_coding/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
index 8049436..addb717 100644
--- a/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/test/TwoWayCommunication.cc
@@ -34,14 +34,16 @@
#define MAX_FILE_NAME_LENGTH_BYTE 500
TwoWayCommunication::TwoWayCommunication(int testMode)
- : _acmA(AudioCodingModule::Create()),
- _acmRefA(AudioCodingModule::Create()),
+ : _acmA(AudioCodingModule::Create(1)),
+ _acmRefA(AudioCodingModule::Create(3)),
_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));
}
@@ -60,7 +62,7 @@
void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
uint8_t* codecID_B) {
- std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create());
+ std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
uint8_t noCodec = tmpACM->NumberOfCodecs();
CodecInst codecInst;
printf("List of Supported Codecs\n");
diff --git a/modules/audio_coding/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc
index 407f709..3f78ea6 100644
--- a/modules/audio_coding/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -64,8 +64,8 @@
class DelayTest {
public:
DelayTest()
- : acm_a_(AudioCodingModule::Create()),
- acm_b_(AudioCodingModule::Create()),
+ : acm_a_(AudioCodingModule::Create(0)),
+ acm_b_(AudioCodingModule::Create(1)),
channel_a2b_(new Channel),
test_cntr_(0),
encoding_sample_rate_hz_(8000) {}
diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
index a44259f..a14f795 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -67,8 +67,8 @@
}
ISACTest::ISACTest(int testMode)
- : _acmA(AudioCodingModule::Create()),
- _acmB(AudioCodingModule::Create()),
+ : _acmA(AudioCodingModule::Create(1)),
+ _acmB(AudioCodingModule::Create(2)),
_testMode(testMode) {}
ISACTest::~ISACTest() {}
diff --git a/modules/audio_coding/test/insert_packet_with_timing.cc b/modules/audio_coding/test/insert_packet_with_timing.cc
index 2c0e54b..500375c 100644
--- a/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -61,8 +61,8 @@
InsertPacketWithTiming()
: sender_clock_(new SimulatedClock(0)),
receiver_clock_(new SimulatedClock(0)),
- send_acm_(AudioCodingModule::Create(sender_clock_)),
- receive_acm_(AudioCodingModule::Create(receiver_clock_)),
+ send_acm_(AudioCodingModule::Create(0, sender_clock_)),
+ receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
channel_(new Channel),
seq_num_fid_(fopen(FLAG_seq_num, "rt")),
send_ts_fid_(fopen(FLAG_send_ts, "rt")),
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index b7acc0f..7b54668 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -27,7 +27,7 @@
namespace webrtc {
OpusTest::OpusTest()
- : acm_receiver_(AudioCodingModule::Create()),
+ : acm_receiver_(AudioCodingModule::Create(0)),
channel_a2b_(NULL),
counter_(0),
payload_type_(255),
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 03135da..2a75706 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -22,7 +22,7 @@
class TargetDelayTest : public ::testing::Test {
protected:
- TargetDelayTest() : acm_(AudioCodingModule::Create()) {}
+ TargetDelayTest() : acm_(AudioCodingModule::Create(0)) {}
~TargetDelayTest() {}