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Support receiving DTMF for multiple RTP clock rates.

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BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2337473002
Cr-Commit-Position: refs/heads/master@{#15128}
This commit is contained in:
solenberg
2016-11-17 04:45:19 -08:00
committed by Commit bot
parent fbfb536ee9
commit 2779bab02a
16 changed files with 266 additions and 93 deletions
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85
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
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@ -173,6 +173,52 @@ class NetEqImplTest : public ::testing::Test {
}
}
void TestDtmfPacket(NetEqDecoder decoder_type) {
const size_t kPayloadLength = 4;
const uint8_t kPayloadType = 110;
const uint32_t kReceiveTime = 17;
const int kSampleRateHz = 16000;
config_.sample_rate_hz = kSampleRateHz;
UseNoMocks();
CreateInstance();
// Event: 2, E bit, Volume: 17, Length: 4336.
uint8_t payload[kPayloadLength] = { 0x02, 0x80 + 0x11, 0x10, 0xF0 };
WebRtcRTPHeader rtp_header;
rtp_header.header.payloadType = kPayloadType;
rtp_header.header.sequenceNumber = 0x1234;
rtp_header.header.timestamp = 0x12345678;
rtp_header.header.ssrc = 0x87654321;
EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
decoder_type, "telephone-event", kPayloadType));
// Insert first packet.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Pull audio once.
const size_t kMaxOutputSize =
static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_FALSE(muted);
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Verify first 64 samples of actual output.
const std::vector<int16_t> kOutput({
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, -1578, -2816, -3460, -3403, -2709, -1594,
-363, 671, 1269, 1328, 908, 202, -513, -964, -955, -431, 504, 1617,
2602, 3164, 3101, 2364, 1073, -511, -2047, -3198, -3721, -3525, -2688,
-1440, -99, 1015, 1663, 1744, 1319, 588, -171, -680, -747, -315, 515,
1512, 2378, 2828, 2674, 1877, 568, -986, -2446, -3482, -3864, -3516,
-2534, -1163 });
ASSERT_GE(kMaxOutputSize, kOutput.size());
EXPECT_TRUE(std::equal(kOutput.begin(), kOutput.end(), output.data_));
}
std::unique_ptr<NetEqImpl> neteq_;
NetEq::Config config_;
TickTimer* tick_timer_ = nullptr;
@ -385,37 +431,20 @@ TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
EXPECT_EQ(rtp_header.header.sequenceNumber, test_packet->sequence_number);
}
TEST_F(NetEqImplTest, TestDtmfPacket) {
UseNoMocks();
CreateInstance();
const size_t kPayloadLength = 4;
const uint8_t kPayloadType = 110;
const uint32_t kReceiveTime = 17;
const int kSampleRateHz = 8000;
// Event: 2, E bit, Volume: 63, Length: 4176.
uint8_t payload[kPayloadLength] = { 0x02, 0x80 + 0x3F, 0x10, 0xF0 };
WebRtcRTPHeader rtp_header;
rtp_header.header.payloadType = kPayloadType;
rtp_header.header.sequenceNumber = 0x1234;
rtp_header.header.timestamp = 0x12345678;
rtp_header.header.ssrc = 0x87654321;
TEST_F(NetEqImplTest, TestDtmfPacketAVT) {
TestDtmfPacket(NetEqDecoder::kDecoderAVT);
}
EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
NetEqDecoder::kDecoderAVT, "telephone-event", kPayloadType));
TEST_F(NetEqImplTest, TestDtmfPacketAVT16kHz) {
TestDtmfPacket(NetEqDecoder::kDecoderAVT16kHz);
}
// Insert one packet.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
TEST_F(NetEqImplTest, TestDtmfPacketAVT32kHz) {
TestDtmfPacket(NetEqDecoder::kDecoderAVT32kHz);
}
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_FALSE(muted);
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
TEST_F(NetEqImplTest, TestDtmfPacketAVT48kHz) {
TestDtmfPacket(NetEqDecoder::kDecoderAVT48kHz);
}
// This test verifies that timestamps propagate from the incoming packets