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

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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

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),