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Fix bug in DTMF generation where events with level > 36 would be ignored.

Browse Source 
BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2404183003
Cr-Commit-Position: refs/heads/master@{#14598}
This commit is contained in:
solenberg 2016-10-11 04:35:34 -07:00 committed by Commit bot
parent 113bdcadf3
commit 99df6c03c3
6 changed files with 68 additions and 14 deletions
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2
webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
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@ -99,7 +99,7 @@ int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp,
// existing one.
int DtmfBuffer::InsertEvent(const DtmfEvent& event) {
if (event.event_no < 0 || event.event_no > 15 ||
event.volume < 0 || event.volume > 36 ||
event.volume < 0 || event.volume > 63 ||
event.duration <= 0 || event.duration > 65535) {
LOG(LS_WARNING) << "InsertEvent invalid parameters";
return kInvalidEventParameters;

2
webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
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@ -282,7 +282,7 @@ TEST(DtmfBuffer, InvalidEvents) {
// Invalid volume.
event.volume = -1;
EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
event.volume = 37;
event.volume = 64;
EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
event.volume = 0; // Valid value;

37
webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
View File

@ -30,7 +30,8 @@
#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
#include <assert.h>
#include "webrtc/base/arraysize.h"
#include "webrtc/base/checks.h"
namespace webrtc {
@ -86,12 +87,16 @@ const int DtmfToneGenerator::kInitValue2[4][16] = {
{ 2851, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 3148,
3476, 3476, 3476, 3476} };
// Amplitude multipliers for volume values 0 through 36, corresponding to
// 0 dBm0 through -36 dBm0. Values are in Q14.
const int DtmfToneGenerator::kAmplitude[37] = {
// Amplitude multipliers for volume values 0 through 63, corresponding to
// 0 dBm0 through -63 dBm0. Values are in Q14.
// for a in range(0, 64):
// print round(16141.0 * 10**(-float(a)/20))
const int DtmfToneGenerator::kAmplitude[64] = {
16141, 14386, 12821, 11427, 10184, 9077, 8090, 7210, 6426, 5727, 5104, 4549,
4054, 3614, 3221, 2870, 2558, 2280, 2032, 1811, 1614, 1439, 1282, 1143,
1018, 908, 809, 721, 643, 573, 510, 455, 405, 361, 322, 287, 256 };
1018, 908, 809, 721, 643, 573, 510, 455, 405, 361, 322, 287, 256, 228, 203,
181, 161, 144, 128, 114, 102, 91, 81, 72, 64, 57, 51, 45, 41, 36, 32, 29,
26, 23, 20, 18, 16, 14, 13, 11 };
// Constructor.
DtmfToneGenerator::DtmfToneGenerator()
@ -106,7 +111,7 @@ DtmfToneGenerator::DtmfToneGenerator()
// Returns 0 on success, otherwise an error code.
int DtmfToneGenerator::Init(int fs, int event, int attenuation) {
initialized_ = false;
int fs_index;
size_t fs_index;
if (fs == 8000) {
fs_index = 0;
} else if (fs == 16000) {
@ -116,7 +121,7 @@ int DtmfToneGenerator::Init(int fs, int event, int attenuation) {
} else if (fs == 48000) {
fs_index = 3;
} else {
assert(false);
RTC_NOTREACHED();
fs_index = 1; // Default to 8000 Hz.
}
@ -124,16 +129,32 @@ int DtmfToneGenerator::Init(int fs, int event, int attenuation) {
return kParameterError; // Invalid event number.
}
if (attenuation < 0 || attenuation > 36) {
if (attenuation < 0 || attenuation > 63) {
return kParameterError; // Invalid attenuation.
}
// Look up oscillator coefficient for low and high frequencies.
RTC_DCHECK_LE(0u, fs_index);
RTC_DCHECK_GT(arraysize(kCoeff1), fs_index);
RTC_DCHECK_GT(arraysize(kCoeff2), fs_index);
RTC_DCHECK_LE(0, event);
RTC_DCHECK_GT(arraysize(kCoeff1[fs_index]), static_cast<size_t>(event));
RTC_DCHECK_GT(arraysize(kCoeff2[fs_index]), static_cast<size_t>(event));
coeff1_ = kCoeff1[fs_index][event];
coeff2_ = kCoeff2[fs_index][event];
// Look up amplitude multiplier.
RTC_DCHECK_LE(0, attenuation);
RTC_DCHECK_GT(arraysize(kAmplitude), static_cast<size_t>(attenuation));
amplitude_ = kAmplitude[attenuation];
// Initialize sample history.
RTC_DCHECK_LE(0u, fs_index);
RTC_DCHECK_GT(arraysize(kInitValue1), fs_index);
RTC_DCHECK_GT(arraysize(kInitValue2), fs_index);
RTC_DCHECK_LE(0, event);
RTC_DCHECK_GT(arraysize(kInitValue1[fs_index]), static_cast<size_t>(event));
RTC_DCHECK_GT(arraysize(kInitValue2[fs_index]), static_cast<size_t>(event));
sample_history1_[0] = kInitValue1[fs_index][event];
sample_history1_[1] = 0;
sample_history2_[0] = kInitValue2[fs_index][event];

2
webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
View File

@ -38,7 +38,7 @@ class DtmfToneGenerator {
static const int kCoeff2[4][16]; // 2nd oscillator model coefficient table.
static const int kInitValue1[4][16]; // Initialization for 1st oscillator.
static const int kInitValue2[4][16]; // Initialization for 2nd oscillator.
static const int kAmplitude[37]; // Amplitude for 0 through -36 dBm0.
static const int kAmplitude[64]; // Amplitude for 0 through -63 dBm0.
static const int16_t kAmpMultiplier = 23171; // 3 dB attenuation (in Q15).
bool initialized_; // True if generator is initialized properly.

6
webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
View File

@ -70,7 +70,7 @@ class DtmfToneGeneratorTest : public ::testing::Test {
ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, 0)); // 0 attenuation.
EXPECT_EQ(kNumSamples, tone_gen_.Generate(kNumSamples, &ref_signal));
// Test every 5 steps (to save time).
for (int attenuation = 1; attenuation <= 36; attenuation += 5) {
for (int attenuation = 1; attenuation <= 63; attenuation += 5) {
std::ostringstream ss;
ss << "Checking event " << event << " at sample rate " << fs_hz;
ss << "; attenuation " << attenuation;
@ -164,8 +164,8 @@ TEST(DtmfToneGenerator, TestErrors) {
tone_gen.Init(fs, 16, attenuation));
// Initialize with invalid attenuation -1.
EXPECT_EQ(DtmfToneGenerator::kParameterError, tone_gen.Init(fs, event, -1));
// Initialize with invalid attenuation 37.
EXPECT_EQ(DtmfToneGenerator::kParameterError, tone_gen.Init(fs, event, 37));
// Initialize with invalid attenuation 64.
EXPECT_EQ(DtmfToneGenerator::kParameterError, tone_gen.Init(fs, event, 64));
EXPECT_FALSE(tone_gen.initialized()); // Should still be uninitialized.
// Initialize with valid parameters.

33
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -381,6 +381,39 @@ TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
EXPECT_EQ(rtp_header.header.sequenceNumber, test_header->sequenceNumber);
}
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;
EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
NetEqDecoder::kDecoderAVT, "telephone-event", kPayloadType));
// Insert one 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_);
}
// This test verifies that timestamps propagate from the incoming packets
// through to the sync buffer and to the playout timestamp.
TEST_F(NetEqImplTest, VerifyTimestampPropagation) {