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NetEq: only update current_rtp_payload_type_ when validated

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The current_rtp_payload_type_ should only be updated when the packet is
actually inserted into the packet buffer, since then the payload type
has been validated. This CL removes an unvalidated setting of this value
that happened after SSRC change or upon first packet.

BUG=webrtc:5447

Review-Url: https://codereview.webrtc.org/2270793003
Cr-Commit-Position: refs/heads/master@{#13910}
This commit is contained in:
henrik.lundin
2016-08-25 00:44:24 -07:00
committed by Commit bot
parent fe8f48962a
commit 549d80b979
2 changed files with 9 additions and 10 deletions
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7
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
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@ -1521,7 +1521,10 @@ TEST_F(NetEqDecodingTest, CngFirst) {
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
// Insert some speech packets.
for (int i = 0; i < 3; ++i) {
const uint32_t first_speech_timestamp = timestamp;
int timeout_counter = 0;
do {
ASSERT_LT(timeout_counter++, 20) << "Test timed out";
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
++seq_no;
@ -1530,7 +1533,7 @@ TEST_F(NetEqDecodingTest, CngFirst) {
// Pull audio once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
} while (!IsNewerTimestamp(out_frame_.timestamp_, first_speech_timestamp));
// Verify speech output.
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
}