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Fixing a bug in ACM2 where the output frame energy was incorrectly set

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The value of AudioFrame::energy_ must be set to -1 in order to have the
energy calculated later on in the AudioConferenceMixer module. This was
not the case in ACM2, where the value was set to 0 instead. This
resulted in bad audio for multi-party calls (5 or more participants).

Implemented a unit test to verify ACM output frame.

BUG=3255
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5969 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-04-24 08:11:39 +00:00
parent f26c9e8369
commit 0a2277448e
2 changed files with 21 additions and 3 deletions
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4
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -1707,7 +1707,9 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
}
audio_frame->id_ = id_;
audio_frame->energy_ = 0;
// The energy must be -1 in order to have the energy calculated later on in
// the AudioConferenceMixer module.
audio_frame->energy_ = static_cast<uint32_t>(-1);
audio_frame->timestamp_ = 0;
return 0;
}

20
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
View File

@ -58,8 +58,9 @@ class RtpUtility {
class AudioCodingModuleTest : public ::testing::Test {
protected:
AudioCodingModuleTest()
: rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
acm2_(AudioCodingModule::Create(1 /*id*/)) {}
: id_(1),
rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
acm2_(AudioCodingModule::Create(id_)) {}
~AudioCodingModuleTest() {}
@ -92,6 +93,7 @@ class AudioCodingModuleTest : public ::testing::Test {
ASSERT_EQ(0, acm2_->PlayoutData10Ms(-1, &audio_frame));
}
const int id_;
scoped_ptr<RtpUtility> rtp_utility_;
scoped_ptr<AudioCodingModule> acm2_;
@ -169,4 +171,18 @@ TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(NetEqCalls)) {
EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
}
TEST_F(AudioCodingModuleTest, VerifyOutputFrame) {
AudioFrame audio_frame;
const int kSampleRateHz = 32000;
EXPECT_EQ(0, acm2_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
// The energy must be -1 in order to have the energy calculated later on in
// the AudioConferenceMixer module.
EXPECT_EQ(static_cast<uint32_t>(-1), audio_frame.energy_);
EXPECT_EQ(id_, audio_frame.id_);
EXPECT_EQ(0u, audio_frame.timestamp_);
EXPECT_GT(audio_frame.num_channels_, 0);
EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);
EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
}
} // namespace webrtc