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}
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 6cfe464..082506a 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -110,7 +110,6 @@
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 d5f196b..307c906 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(0, &clock_)),
+ acm_(webrtc::AudioCodingModule::Create(&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 5997d12..c48fbef 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -269,7 +269,6 @@
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_);
@@ -456,8 +455,7 @@
AudioCodingModuleImpl::AudioCodingModuleImpl(
const AudioCodingModule::Config& config)
- : id_(config.id),
- expected_codec_ts_(0xD87F3F9F),
+ : expected_codec_ts_(0xD87F3F9F),
expected_in_ts_(0xD87F3F9F),
receiver_(config),
bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
@@ -1120,7 +1118,6 @@
LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
return -1;
}
- audio_frame->id_ = id_;
return 0;
}
@@ -1286,7 +1283,7 @@
} // namespace
AudioCodingModule::Config::Config()
- : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {
+ : 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;
@@ -1296,17 +1293,15 @@
AudioCodingModule::Config::~Config() = default;
// Create module
-AudioCodingModule* AudioCodingModule::Create(int id) {
+AudioCodingModule* AudioCodingModule::Create() {
Config config;
- config.id = id;
config.clock = Clock::GetRealTimeClock();
config.decoder_factory = CreateBuiltinAudioDecoderFactory();
return Create(config);
}
-AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
+AudioCodingModule* AudioCodingModule::Create(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 80fc4d8..a010619 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -157,8 +157,7 @@
class AudioCodingModuleTestOldApi : public ::testing::Test {
protected:
AudioCodingModuleTestOldApi()
- : id_(1),
- rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
+ : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
clock_(Clock::GetRealTimeClock()) {}
~AudioCodingModuleTestOldApi() {}
@@ -166,7 +165,7 @@
void TearDown() {}
void SetUp() {
- acm_.reset(AudioCodingModule::Create(id_, clock_));
+ acm_.reset(AudioCodingModule::Create(clock_));
rtp_utility_->Populate(&rtp_header_);
@@ -230,7 +229,6 @@
VerifyEncoding();
}
- const int id_;
std::unique_ptr<RtpUtility> rtp_utility_;
std::unique_ptr<AudioCodingModule> acm_;
PacketizationCallbackStubOldApi packet_cb_;
@@ -314,7 +312,6 @@
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),