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Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ )

Browse Source 
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}
This commit is contained in:
solenberg
2017-09-27 11:28:14 -07:00
committed by Commit Bot
parent 1c46a35c5e
commit e423a9de93
25 changed files with 90 additions and 52 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
modules/audio_coding/acm2/acm_receive_test.cc
View File

@ -110,6 +110,7 @@ AudioCodingModule::Config MakeAcmConfig(
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;

2
modules/audio_coding/acm2/acm_send_test.cc
View File

@ -28,7 +28,7 @@ AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
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_(

13
modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -269,6 +269,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
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 @@ void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
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 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
return -1;
}
audio_frame->id_ = id_;
return 0;
}
@ -1283,7 +1286,7 @@ ANAStats AudioCodingModuleImpl::GetANAStats() const {
} // 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(const Config&) = default;
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);

7
modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -157,7 +157,8 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
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 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
void TearDown() {}
void SetUp() {
acm_.reset(AudioCodingModule::Create(clock_));
acm_.reset(AudioCodingModule::Create(id_, clock_));
rtp_utility_->Populate(&rtp_header_);
@ -229,6 +230,7 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
VerifyEncoding();
}
const int id_;
std::unique_ptr<RtpUtility> rtp_utility_;
std::unique_ptr<AudioCodingModule> acm_;
PacketizationCallbackStubOldApi packet_cb_;
@ -312,6 +314,7 @@ TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
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),

5
modules/audio_coding/include/audio_coding_module.h
View File

@ -70,6 +70,7 @@ class AudioCodingModule {
Config(const Config&);
~Config();
int id;
NetEq::Config neteq_config;
Clock* clock;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
@ -82,8 +83,8 @@ class AudioCodingModule {
// 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;

4
modules/audio_coding/test/APITest.cc
View File

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

4
modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -281,7 +281,7 @@ void EncodeDecodeTest::Perform() {
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 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
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");

2
modules/audio_coding/test/PacketLossTest.cc
View File

@ -127,7 +127,7 @@ void PacketLossTest::Perform() {
#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_);

4
modules/audio_coding/test/TestAllCodecs.cc
View File

@ -104,8 +104,8 @@ void TestPack::reset_payload_size() {
}
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),

4
modules/audio_coding/test/TestRedFec.cc
View File

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

4
modules/audio_coding/test/TestStereo.cc
View File

@ -108,8 +108,8 @@ void TestPackStereo::set_lost_packet(bool lost) {
}
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),

4
modules/audio_coding/test/TestVADDTX.cc
View File

@ -62,8 +62,8 @@ void ActivityMonitor::GetStatistics(uint32_t* counter) {
}
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()));

8
modules/audio_coding/test/TwoWayCommunication.cc
View File

@ -34,14 +34,16 @@ namespace webrtc {
#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 @@ TwoWayCommunication::~TwoWayCommunication() {
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");

4
modules/audio_coding/test/delay_test.cc
View File

@ -64,8 +64,8 @@ struct TestSettings {
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) {}

4
modules/audio_coding/test/iSACTest.cc
View File

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

4
modules/audio_coding/test/insert_packet_with_timing.cc
View File

@ -61,8 +61,8 @@ class InsertPacketWithTiming {
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")),

2
modules/audio_coding/test/opus_test.cc
View File

@ -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),

2
modules/audio_coding/test/target_delay_unittest.cc
View File

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