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Remove unused method OutputMixer::PlayDtmfTone() and infrastructure.

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BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1796183002

Cr-Commit-Position: refs/heads/master@{#11990}
This commit is contained in:
solenberg 2016-03-14 15:32:50 -07:00 committed by Commit bot
parent c4ec4a2e51
commit e50872be13
7 changed files with 1 additions and 563 deletions
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2
webrtc/voice_engine/BUILD.gn
View File

@ -16,8 +16,6 @@ source_set("voice_engine") {
"channel_manager.h",
"channel_proxy.cc",
"channel_proxy.h",
"dtmf_inband.cc",
"dtmf_inband.h",
"include/voe_audio_processing.h",
"include/voe_base.h",
"include/voe_codec.h",

389
webrtc/voice_engine/dtmf_inband.cc
View File

@ -1,389 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/voice_engine/dtmf_inband.h"
#include <assert.h>
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {
const int16_t Dtmf_a_times2Tab8Khz[8]=
{
27978, 26956, 25701, 24219,
19073, 16325, 13085, 9314
};
const int16_t Dtmf_a_times2Tab16Khz[8]=
{
31548, 31281, 30951, 30556,
29144, 28361, 27409, 26258
};
const int16_t Dtmf_a_times2Tab32Khz[8]=
{
32462,32394, 32311, 32210, 31849, 31647, 31400, 31098
};
// Second table is sin(2*pi*f/fs) in Q14
const int16_t Dtmf_ym2Tab8Khz[8]=
{
8527, 9315, 10163, 11036,
13322, 14206, 15021, 15708
};
const int16_t Dtmf_ym2Tab16Khz[8]=
{
4429, 4879, 5380, 5918,
7490, 8207, 8979, 9801
};
const int16_t Dtmf_ym2Tab32Khz[8]=
{
2235, 2468, 2728, 3010, 3853, 4249, 4685, 5164
};
const int16_t Dtmf_dBm0kHz[37]=
{
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
};
DtmfInband::DtmfInband(int32_t id) :
_id(id),
_outputFrequencyHz(8000),
_frameLengthSamples(0),
_remainingSamples(0),
_eventCode(0),
_attenuationDb(0),
_lengthMs(0),
_reinit(true),
_playing(false),
_delaySinceLastToneMS(1000)
{
memset(_oldOutputLow, 0, sizeof(_oldOutputLow));
memset(_oldOutputHigh, 0, sizeof(_oldOutputHigh));
}
DtmfInband::~DtmfInband()
{
}
int
DtmfInband::SetSampleRate(uint16_t frequency)
{
if (frequency != 8000 &&
frequency != 16000 &&
frequency != 32000)
{
// invalid sample rate
assert(false);
return -1;
}
_outputFrequencyHz = frequency;
return 0;
}
int
DtmfInband::GetSampleRate(uint16_t& frequency)
{
frequency = _outputFrequencyHz;
return 0;
}
void
DtmfInband::Init()
{
_remainingSamples = 0;
_frameLengthSamples = 0;
_eventCode = 0;
_attenuationDb = 0;
_lengthMs = 0;
_reinit = true;
_oldOutputLow[0] = 0;
_oldOutputLow[1] = 0;
_oldOutputHigh[0] = 0;
_oldOutputHigh[1] = 0;
_delaySinceLastToneMS = 1000;
}
int
DtmfInband::AddTone(uint8_t eventCode,
int32_t lengthMs,
int32_t attenuationDb)
{
rtc::CritScope lock(&_critSect);
if (attenuationDb > 36 || eventCode > 15)
{
assert(false);
return -1;
}
if (IsAddingTone())
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_id,-1),
"DtmfInband::AddTone() new tone interrupts ongoing tone");
}
ReInit();
_frameLengthSamples = static_cast<int16_t> (_outputFrequencyHz / 100);
_eventCode = static_cast<int16_t> (eventCode);
_attenuationDb = static_cast<int16_t> (attenuationDb);
_remainingSamples = static_cast<int32_t>
(lengthMs * (_outputFrequencyHz / 1000));
_lengthMs = lengthMs;
return 0;
}
int
DtmfInband::ResetTone()
{
rtc::CritScope lock(&_critSect);
ReInit();
_frameLengthSamples = static_cast<int16_t> (_outputFrequencyHz / 100);
_remainingSamples = static_cast<int32_t>
(_lengthMs * (_outputFrequencyHz / 1000));
return 0;
}
int
DtmfInband::StartTone(uint8_t eventCode,
int32_t attenuationDb)
{
rtc::CritScope lock(&_critSect);
if (attenuationDb > 36 || eventCode > 15)
{
assert(false);
return -1;
}
if (IsAddingTone())
{
return -1;
}
ReInit();
_frameLengthSamples = static_cast<int16_t> (_outputFrequencyHz / 100);
_eventCode = static_cast<int16_t> (eventCode);
_attenuationDb = static_cast<int16_t> (attenuationDb);
_playing = true;
return 0;
}
int
DtmfInband::StopTone()
{
rtc::CritScope lock(&_critSect);
if (!_playing)
{
return 0;
}
_playing = false;
return 0;
}
// Shall be called between tones
void
DtmfInband::ReInit()
{
_reinit = true;
}
bool
DtmfInband::IsAddingTone()
{
rtc::CritScope lock(&_critSect);
return (_remainingSamples > 0 || _playing);
}
int
DtmfInband::Get10msTone(int16_t output[320],
uint16_t& outputSizeInSamples)
{
rtc::CritScope lock(&_critSect);
if (DtmfFix_generate(output,
_eventCode,
_attenuationDb,
_frameLengthSamples,
_outputFrequencyHz) == -1)
{
return -1;
}
_remainingSamples -= _frameLengthSamples;
outputSizeInSamples = _frameLengthSamples;
_delaySinceLastToneMS = 0;
return 0;
}
void
DtmfInband::UpdateDelaySinceLastTone()
{
rtc::CritScope lock(&_critSect);
_delaySinceLastToneMS += kDtmfFrameSizeMs;
// avoid wraparound
if (_delaySinceLastToneMS > (1<<30))
{
_delaySinceLastToneMS = 1000;
}
}
uint32_t
DtmfInband::DelaySinceLastTone() const
{
rtc::CritScope lock(&_critSect);
return _delaySinceLastToneMS;
}
int16_t
DtmfInband::DtmfFix_generate(int16_t *decoded,
int16_t value,
int16_t volume,
int16_t frameLen,
int16_t fs)
{
const int16_t *a_times2Tbl;
const int16_t *y2_Table;
int16_t a1_times2 = 0, a2_times2 = 0;
if (fs==8000) {
a_times2Tbl=Dtmf_a_times2Tab8Khz;
y2_Table=Dtmf_ym2Tab8Khz;
} else if (fs==16000) {
a_times2Tbl=Dtmf_a_times2Tab16Khz;
y2_Table=Dtmf_ym2Tab16Khz;
} else if (fs==32000) {
a_times2Tbl=Dtmf_a_times2Tab32Khz;
y2_Table=Dtmf_ym2Tab32Khz;
} else {
return(-1);
}
if ((value==1)||(value==2)||(value==3)||(value==12)) {
a1_times2=a_times2Tbl[0];
if (_reinit) {
_oldOutputLow[0]=y2_Table[0];
_oldOutputLow[1]=0;
}
} else if ((value==4)||(value==5)||(value==6)||(value==13)) {
a1_times2=a_times2Tbl[1];
if (_reinit) {
_oldOutputLow[0]=y2_Table[1];
_oldOutputLow[1]=0;
}
} else if ((value==7)||(value==8)||(value==9)||(value==14)) {
a1_times2=a_times2Tbl[2];
if (_reinit) {
_oldOutputLow[0]=y2_Table[2];
_oldOutputLow[1]=0;
}
} else if ((value==10)||(value==0)||(value==11)||(value==15)) {
a1_times2=a_times2Tbl[3];
if (_reinit) {
_oldOutputLow[0]=y2_Table[3];
_oldOutputLow[1]=0;
}
}
if ((value==1)||(value==4)||(value==7)||(value==10)) {
a2_times2=a_times2Tbl[4];
if (_reinit) {
_oldOutputHigh[0]=y2_Table[4];
_oldOutputHigh[1]=0;
_reinit=false;
}
} else if ((value==2)||(value==5)||(value==8)||(value==0)) {
a2_times2=a_times2Tbl[5];
if (_reinit) {
_oldOutputHigh[0]=y2_Table[5];
_oldOutputHigh[1]=0;
_reinit=false;
}
} else if ((value==3)||(value==6)||(value==9)||(value==11)) {
a2_times2=a_times2Tbl[6];
if (_reinit) {
_oldOutputHigh[0]=y2_Table[6];
_oldOutputHigh[1]=0;
_reinit=false;
}
} else if ((value==12)||(value==13)||(value==14)||(value==15)) {
a2_times2=a_times2Tbl[7];
if (_reinit) {
_oldOutputHigh[0]=y2_Table[7];
_oldOutputHigh[1]=0;
_reinit=false;
}
}
return (DtmfFix_generateSignal(a1_times2,
a2_times2,
volume,
decoded,
frameLen));
}
int16_t
DtmfInband::DtmfFix_generateSignal(int16_t a1_times2,
int16_t a2_times2,
int16_t volume,
int16_t *signal,
int16_t length)
{
int i;
/* Generate Signal */
for (i=0;i<length;i++) {
int32_t tempVal;
int16_t tempValLow, tempValHigh;
/* Use recursion formula y[n] = a*2*y[n-1] - y[n-2] */
tempValLow = (int16_t)(((( (int32_t)(a1_times2 *
_oldOutputLow[1])) + 8192) >> 14) - _oldOutputLow[0]);
tempValHigh = (int16_t)(((( (int32_t)(a2_times2 *
_oldOutputHigh[1])) + 8192) >> 14) - _oldOutputHigh[0]);
/* Update memory */
_oldOutputLow[0]=_oldOutputLow[1];
_oldOutputLow[1]=tempValLow;
_oldOutputHigh[0]=_oldOutputHigh[1];
_oldOutputHigh[1]=tempValHigh;
tempVal = (int32_t)(kDtmfAmpLow * tempValLow) +
(int32_t)(kDtmfAmpHigh * tempValHigh);
/* Norm the signal to Q14 */
tempVal=(tempVal+16384)>>15;
/* Scale the signal to correct dbM0 value */
signal[i]=(int16_t)((tempVal*Dtmf_dBm0kHz[volume]+8192)>>14);
}
return(0);
}
} // namespace webrtc

90
webrtc/voice_engine/dtmf_inband.h
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@ -1,90 +0,0 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_H_
#define WEBRTC_VOICE_ENGINE_DTMF_INBAND_H_
#include "webrtc/typedefs.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
#include "webrtc/base/criticalsection.h"
namespace webrtc {
// TODO(solenberg): Used as a DTMF tone generator in voe::OutputMixer. Pull out
// the one in NetEq and use that instead? We don't need several
// implemenations of this.
class DtmfInband
{
public:
DtmfInband(int32_t id);
virtual ~DtmfInband();
void Init();
int SetSampleRate(uint16_t frequency);
int GetSampleRate(uint16_t& frequency);
int AddTone(uint8_t eventCode,
int32_t lengthMs,
int32_t attenuationDb);
int ResetTone();
int StartTone(uint8_t eventCode, int32_t attenuationDb);
int StopTone();
bool IsAddingTone();
int Get10msTone(int16_t output[320], uint16_t& outputSizeInSamples);
uint32_t DelaySinceLastTone() const;
void UpdateDelaySinceLastTone();
private:
void ReInit();
int16_t DtmfFix_generate(int16_t* decoded,
int16_t value,
int16_t volume,
int16_t frameLen,
int16_t fs);
private:
enum {kDtmfFrameSizeMs = 10};
enum {kDtmfAmpHigh = 32768};
enum {kDtmfAmpLow = 23171}; // 3 dB lower than the high frequency
int16_t DtmfFix_generateSignal(int16_t a1_times2,
int16_t a2_times2,
int16_t volume,
int16_t* signal,
int16_t length);
private:
rtc::CriticalSection _critSect;
int32_t _id;
uint16_t _outputFrequencyHz; // {8000, 16000, 32000}
int16_t _oldOutputLow[2]; // Data needed for oscillator model
int16_t _oldOutputHigh[2]; // Data needed for oscillator model
int16_t _frameLengthSamples; // {80, 160, 320}
int32_t _remainingSamples;
int16_t _eventCode; // [0, 15]
int16_t _attenuationDb; // [0, 36]
int32_t _lengthMs;
bool _reinit; // 'true' if the oscillator should be reinit for next event
bool _playing;
uint32_t _delaySinceLastToneMS; // time since last generated tone [ms]
};
} // namespace webrtc
#endif // #ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_H_

2
webrtc/voice_engine/include/voe_errors.h
View File

@ -47,7 +47,7 @@
#define VE_NO_GQOS 8033
#define VE_INVALID_TIMESTAMP 8034
#define VE_RECEIVE_PACKET_TIMEOUT 8035
#define VE_STILL_PLAYING_PREV_DTMF 8036
// 8036 is not used
#define VE_INIT_FAILED_WRONG_EXPIRY 8037
#define VE_SENDING 8038
#define VE_ENABLE_IPV6_FAILED 8039

73
webrtc/voice_engine/output_mixer.cc
View File

@ -93,7 +93,6 @@ OutputMixer::Create(OutputMixer*& mixer, uint32_t instanceId)
OutputMixer::OutputMixer(uint32_t instanceId) :
_mixerModule(*AudioConferenceMixer::Create(instanceId)),
_audioLevel(),
_dtmfGenerator(instanceId),
_instanceId(instanceId),
_externalMediaCallbackPtr(NULL),
_externalMedia(false),
@ -112,8 +111,6 @@ OutputMixer::OutputMixer(uint32_t instanceId) :
"OutputMixer::OutputMixer() failed to register mixer"
"callbacks");
}
_dtmfGenerator.Init();
}
void
@ -192,21 +189,6 @@ int OutputMixer::DeRegisterExternalMediaProcessing()
return 0;
}
int OutputMixer::PlayDtmfTone(uint8_t eventCode, int lengthMs,
int attenuationDb)
{
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
"OutputMixer::PlayDtmfTone()");
if (_dtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb) != 0)
{
_engineStatisticsPtr->SetLastError(VE_STILL_PLAYING_PREV_DTMF,
kTraceError,
"OutputMixer::PlayDtmfTone()");
return -1;
}
return 0;
}
int32_t
OutputMixer::SetMixabilityStatus(MixerParticipant& participant,
bool mixable)
@ -492,12 +474,6 @@ OutputMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm)
_mixingFrequencyHz = _audioFrame.sample_rate_hz_;
}
// --- Insert inband Dtmf tone
if (_dtmfGenerator.IsAddingTone())
{
InsertInbandDtmfTone();
}
// Scale left and/or right channel(s) if balance is active
if (_panLeft != 1.0 || _panRight != 1.0)
{
@ -554,54 +530,5 @@ OutputMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm)
return 0;
}
// ----------------------------------------------------------------------------
// Private methods
// ----------------------------------------------------------------------------
int
OutputMixer::InsertInbandDtmfTone()
{
uint16_t sampleRate(0);
_dtmfGenerator.GetSampleRate(sampleRate);
if (sampleRate != _audioFrame.sample_rate_hz_)
{
// Update sample rate of Dtmf tone since the mixing frequency changed.
_dtmfGenerator.SetSampleRate(
(uint16_t)(_audioFrame.sample_rate_hz_));
// Reset the tone to be added taking the new sample rate into account.
_dtmfGenerator.ResetTone();
}
int16_t toneBuffer[320];
uint16_t toneSamples(0);
if (_dtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
"OutputMixer::InsertInbandDtmfTone() inserting Dtmf"
"tone failed");
return -1;
}
// replace mixed audio with Dtmf tone
if (_audioFrame.num_channels_ == 1)
{
// mono
memcpy(_audioFrame.data_, toneBuffer, sizeof(int16_t)
* toneSamples);
} else
{
// stereo
for (size_t i = 0; i < _audioFrame.samples_per_channel_; i++)
{
_audioFrame.data_[2 * i] = toneBuffer[i];
_audioFrame.data_[2 * i + 1] = 0;
}
}
assert(_audioFrame.samples_per_channel_ == toneSamples);
return 0;
}
} // namespace voe
} // namespace webrtc

6
webrtc/voice_engine/output_mixer.h
View File

@ -17,7 +17,6 @@
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/dtmf_inband.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
@ -50,9 +49,6 @@ public:
int DeRegisterExternalMediaProcessing();
// VoEDtmf
int PlayDtmfTone(uint8_t eventCode, int lengthMs, int attenuationDb);
int32_t MixActiveChannels();
int32_t DoOperationsOnCombinedSignal(bool feed_data_to_apm);
@ -102,7 +98,6 @@ public:
private:
OutputMixer(uint32_t instanceId);
int InsertInbandDtmfTone();
// uses
Statistics* _engineStatisticsPtr;
@ -118,7 +113,6 @@ private:
// Converts mixed audio to the audio processing rate.
PushResampler<int16_t> audioproc_resampler_;
AudioLevel _audioLevel; // measures audio level for the combined signal
DtmfInband _dtmfGenerator;
int _instanceId;
VoEMediaProcess* _externalMediaCallbackPtr;
bool _externalMedia;

2
webrtc/voice_engine/voice_engine.gyp
View File

@ -52,8 +52,6 @@
'channel_manager.h',
'channel_proxy.cc',
'channel_proxy.h',
'dtmf_inband.cc',
'dtmf_inband.h',
'level_indicator.cc',
'level_indicator.h',
'monitor_module.cc',