audio_processing/agc: Solved building with AGC_DEBUG + few style changes

webrtc did not build if AGC_DEBUG was turned on. This CL fixes that. Has no impact on performance since it is development/debug code. * Name change to WEBRT_AGC_DEBUG_DUMP * Added build flag agc_debug_dump to .gypi * Added missing "%d" in printf at two places * Some line length related style changes Tested audioproc and modules_unittests with GYP_DEFINES=agc_debug_dump=1 webrtc/build/gyp_webrtc BUG=N/A TESTED=locally and trybots R=aluebs@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7271 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 11:21:39 +00:00
parent 0a2087a711
commit ea29787df0
5 changed files with 120 additions and 74 deletions
--- a/webrtc/modules/audio_processing/agc/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/analog_agc.c
@@ -19,7 +19,7 @@

 #include <assert.h>
 #include <stdlib.h>
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
 #include "webrtc/modules/audio_processing/agc/analog_agc.h"
@@ -139,10 +139,10 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
            L = 8;
        } else
        {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
            fprintf(stt->fpt,
                    "AGC->add_mic, frame %d: Invalid number of samples\n\n",
-                    (stt->fcount + 1));
+                    stt->fcount + 1);
 #endif
            return -1;
        }
@@ -160,10 +160,10 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
            L = 16;
        } else
        {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
            fprintf(stt->fpt,
                    "AGC->add_mic, frame %d: Invalid number of samples\n\n",
-                    (stt->fcount + 1));
+                    stt->fcount + 1);
 #endif
            return -1;
        }
@@ -177,10 +177,10 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
            L = 16;
        } else
        {
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
            fprintf(stt->fpt,
                    "AGC->add_mic, frame %d: Invalid sample rate\n\n",
-                    (stt->fcount + 1));
+                    stt->fcount + 1);
 #endif
            return -1;
        }
@@ -343,7 +343,7 @@ int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples)
    {
        if ((samples != 80) && (samples != 160))
        {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
            fprintf(stt->fpt,
                    "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
                    stt->fcount);
@@ -355,7 +355,7 @@ int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples)
    {
        if ((samples != 160) && (samples != 320))
        {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
            fprintf(stt->fpt,
                    "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
                    stt->fcount);
@@ -367,7 +367,7 @@ int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples)
    {
        if ((samples != 160) && (samples != 320))
        {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
            fprintf(stt->fpt,
                    "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
                    stt->fcount);
@@ -377,7 +377,7 @@ int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples)
        subFrames = 160;
    } else
    {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
        fprintf(stt->fpt,
                "AGC->add_far_end, frame %d: Invalid sample rate\n\n",
                stt->fcount + 1);
@@ -657,10 +657,12 @@ void WebRtcAgc_ZeroCtrl(Agc_t *stt, int32_t *inMicLevel, int32_t *env)
            stt->micVol = *inMicLevel;
        }

-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
        fprintf(stt->fpt,
-                "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold, micVol:\n",
-                stt->fcount, stt->micVol);
+                "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold,"
+                " micVol: %d\n",
+                stt->fcount,
+                stt->micVol);
 #endif

        stt->activeSpeech = 0;
@@ -771,14 +773,18 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,

    if (inMicLevelTmp > stt->maxAnalog)
    {
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n", stt->fcount);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        fprintf(stt->fpt,
+                "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n",
+                stt->fcount);
 #endif
        return -1;
    } else if (inMicLevelTmp < stt->minLevel)
    {
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n", stt->fcount);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        fprintf(stt->fpt,
+                "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n",
+                stt->fcount);
 #endif
        return -1;
    }
@@ -813,9 +819,10 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
 #ifdef MIC_LEVEL_FEEDBACK
        //stt->numBlocksMicLvlSat = 0;
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
        fprintf(stt->fpt,
-                "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual decrease, raise vol\n",
+                "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual"
+                " decrease, raise vol\n",
                stt->fcount);
 #endif
    }
@@ -871,10 +878,11 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
        }
        inMicLevelTmp = stt->micVol;

-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
        fprintf(stt->fpt,
                "\tAGC->ProcessAnalog, frame %d: saturated, micVol = %d\n",
-                stt->fcount, stt->micVol);
+                stt->fcount,
+                stt->micVol);
 #endif

        if (stt->micVol < stt->minOutput)
@@ -1011,10 +1019,13 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
 #ifdef MIC_LEVEL_FEEDBACK
                    //stt->numBlocksMicLvlSat = 0;
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
                    fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure > 2ndUpperLim, micVol = %d, maxLevel = %d\n",
-                            stt->fcount, stt->micVol, stt->maxLevel);
+                            "\tAGC->ProcessAnalog, frame %d: measure >"
+                            " 2ndUpperLim, micVol = %d, maxLevel = %d\n",
+                            stt->fcount,
+                            stt->micVol,
+                            stt->maxLevel);
 #endif
                }
            } else if (stt->Rxx160_LPw32 > stt->upperLimit)
@@ -1054,10 +1065,13 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
 #ifdef MIC_LEVEL_FEEDBACK
                    //stt->numBlocksMicLvlSat = 0;
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
                    fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure > UpperLim, micVol = %d, maxLevel = %d\n",
-                            stt->fcount, stt->micVol, stt->maxLevel);
+                            "\tAGC->ProcessAnalog, frame %d: measure >"
+                            " UpperLim, micVol = %d, maxLevel = %d\n",
+                            stt->fcount,
+                            stt->micVol,
+                            stt->maxLevel);
 #endif
                }
            } else if (stt->Rxx160_LPw32 < stt->lowerSecondaryLimit)
@@ -1113,10 +1127,12 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
                        fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
                    }
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
                    fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure < 2ndLowerLim, micVol = %d\n",
-                            stt->fcount, stt->micVol);
+                            "\tAGC->ProcessAnalog, frame %d: measure <"
+                            " 2ndLowerLim, micVol = %d\n",
+                            stt->fcount,
+                            stt->micVol);
 #endif
                }
            } else if (stt->Rxx160_LPw32 < stt->lowerLimit)
@@ -1172,10 +1188,11 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
                        fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
                    }
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
                    fprintf(stt->fpt,
                            "\tAGC->ProcessAnalog, frame %d: measure < LowerLim, micVol = %d\n",
-                            stt->fcount, stt->micVol);
+                            stt->fcount,
+                            stt->micVol);
 #endif

                }
@@ -1272,9 +1289,10 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
    {
        if ((samples != 80) && (samples != 160))
        {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
            fprintf(stt->fpt,
-                    "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
+                    "AGC->Process, frame %d: Invalid number of samples\n\n",
+                    stt->fcount);
 #endif
            return -1;
        }
@@ -1283,9 +1301,10 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
    {
        if ((samples != 160) && (samples != 320))
        {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
            fprintf(stt->fpt,
-                    "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
+                    "AGC->Process, frame %d: Invalid number of samples\n\n",
+                    stt->fcount);
 #endif
            return -1;
        }
@@ -1294,18 +1313,20 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
    {
        if ((samples != 160) && (samples != 320))
        {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
            fprintf(stt->fpt,
-                    "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
+                    "AGC->Process, frame %d: Invalid number of samples\n\n",
+                    stt->fcount);
 #endif
            return -1;
        }
        subFrames = 160;
    } else
    {
-#ifdef AGC_DEBUG// test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
        fprintf(stt->fpt,
-                "AGC->Process, frame %d: Invalid sample rate\n\n", stt->fcount);
+                "AGC->Process, frame %d: Invalid sample rate\n\n",
+                stt->fcount);
 #endif
        return -1;
    }
@@ -1341,7 +1362,7 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
        }
    }

-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
    stt->fcount++;
 #endif

@@ -1350,8 +1371,10 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
        if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, &in_near[i], &in_near_H[i], &out[i], &out_H[i],
                           stt->fs, stt->lowLevelSignal) == -1)
        {
-#ifdef AGC_DEBUG//test log
-            fprintf(stt->fpt, "AGC->Process, frame %d: Error from DigAGC\n\n", stt->fcount);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+            fprintf(stt->fpt,
+                    "AGC->Process, frame %d: Error from DigAGC\n\n",
+                    stt->fcount);
 #endif
            return -1;
        }
@@ -1364,8 +1387,14 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
                return -1;
            }
        }
-#ifdef AGC_DEBUG//test log
-        fprintf(stt->agcLog, "%5d\t%d\t%d\t%d\n", stt->fcount, inMicLevelTmp, *outMicLevel, stt->maxLevel, stt->micVol);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        fprintf(stt->agcLog,
+                "%5d\t%d\t%d\t%d\t%d\n",
+                stt->fcount,
+                inMicLevelTmp,
+                *outMicLevel,
+                stt->maxLevel,
+                stt->micVol);
 #endif

        /* update queue */
@@ -1441,8 +1470,10 @@ int WebRtcAgc_set_config(void *agcInst, WebRtcAgc_config_t agcConfig)
    if (WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]), stt->compressionGaindB,
                           stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget) == -1)
    {
-#ifdef AGC_DEBUG//test log
-        fprintf(stt->fpt, "AGC->set_config, frame %d: Error from calcGainTable\n\n", stt->fcount);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        fprintf(stt->fpt,
+                "AGC->set_config, frame %d: Error from calcGainTable\n\n",
+                stt->fcount);
 #endif
        return -1;
    }
@@ -1498,7 +1529,7 @@ int WebRtcAgc_Create(void **agcInst)
        return -1;
    }

-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
    stt->fpt = fopen("./agc_test_log.txt", "wt");
    stt->agcLog = fopen("./agc_debug_log.txt", "wt");
    stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
@@ -1515,7 +1546,7 @@ int WebRtcAgc_Free(void *state)
    Agc_t *stt;

    stt = (Agc_t *)state;
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
    fclose(stt->fpt);
    fclose(stt->agcLog);
    fclose(stt->digitalAgc.logFile);
@@ -1553,13 +1584,13 @@ int WebRtcAgc_Init(void *agcInst, int32_t minLevel, int32_t maxLevel,
     *            2 - Digital Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
     *            3 - Fixed Digital Gain [compressionGaindB (default 8 dB)]
     */
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
    stt->fcount = 0;
    fprintf(stt->fpt, "AGC->Init\n");
 #endif
    if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital)
    {
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
        fprintf(stt->fpt, "AGC->Init: error, incorrect mode\n\n");
 #endif
        return -1;
@@ -1616,10 +1647,12 @@ int WebRtcAgc_Init(void *agcInst, int32_t minLevel, int32_t maxLevel,
    stt->numBlocksMicLvlSat = 0;
    stt->micLvlSat = 0;
 #endif
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
    fprintf(stt->fpt,
            "AGC->Init: minLevel = %d, maxAnalog = %d, maxLevel = %d\n",
-            stt->minLevel, stt->maxAnalog, stt->maxLevel);
+            stt->minLevel,
+            stt->maxAnalog,
+            stt->maxLevel);
 #endif

    /* Minimum output volume is 4% higher than the available lowest volume level */
@@ -1687,13 +1720,13 @@ int WebRtcAgc_Init(void *agcInst, int32_t minLevel, int32_t maxLevel,
    /* Only positive values are allowed that are not too large */
    if ((minLevel >= maxLevel) || (maxLevel & 0xFC000000))
    {
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
        fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n");
 #endif
        return -1;
    } else
    {
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
        fprintf(stt->fpt, "\n");
 #endif
        return 0;
--- a/webrtc/modules/audio_processing/agc/analog_agc.h
+++ b/webrtc/modules/audio_processing/agc/analog_agc.h
@@ -15,9 +15,8 @@
 #include "webrtc/modules/audio_processing/agc/include/gain_control.h"
 #include "webrtc/typedefs.h"

-//#define AGC_DEBUG
 //#define MIC_LEVEL_FEEDBACK
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif

@@ -122,10 +121,10 @@ typedef struct
    AgcVad_t            vadMic;
    DigitalAgc_t        digitalAgc;

-#ifdef AGC_DEBUG
-    FILE*               fpt;
-    FILE*               agcLog;
-    int32_t             fcount;
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+    FILE* fpt;
+    FILE* agcLog;
+    int32_t fcount;
 #endif

    int16_t             lowLevelSignal;
--- a/webrtc/modules/audio_processing/agc/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/digital_agc.c
@@ -16,7 +16,7 @@

 #include <assert.h>
 #include <string.h>
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif

@@ -274,7 +274,7 @@ int32_t WebRtcAgc_InitDigital(DigitalAgc_t *stt, int16_t agcMode)
    stt->gain = 65536;
    stt->gatePrevious = 0;
    stt->agcMode = agcMode;
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
    stt->frameCounter = 0;
 #endif

@@ -397,9 +397,14 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
            decay = 0;
        }
    }
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
    stt->frameCounter++;
-    fprintf(stt->logFile, "%5.2f\t%d\t%d\t%d\t", (float)(stt->frameCounter) / 100, logratio, decay, stt->vadNearend.stdLongTerm);
+    fprintf(stt->logFile,
+            "%5.2f\t%d\t%d\t%d\t",
+            (float)(stt->frameCounter) / 100,
+            logratio,
+            decay,
+            stt->vadNearend.stdLongTerm);
 #endif
    // Find max amplitude per sub frame
    // iterate over sub frames
@@ -461,10 +466,15 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
        frac = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 19); // Q12
        tmp32 = WEBRTC_SPL_MUL((stt->gainTable[zeros-1] - stt->gainTable[zeros]), frac);
        gains[k + 1] = stt->gainTable[zeros] + WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
-#ifdef AGC_DEBUG
-        if (k == 0)
-        {
-            fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level, stt->capacitorFast, stt->capacitorSlow, zeros);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        if (k == 0) {
+          fprintf(stt->logFile,
+                  "%d\t%d\t%d\t%d\t%d\n",
+                  env[0],
+                  cur_level,
+                  stt->capacitorFast,
+                  stt->capacitorSlow,
+                  zeros);
        }
 #endif
    }
--- a/webrtc/modules/audio_processing/agc/digital_agc.h
+++ b/webrtc/modules/audio_processing/agc/digital_agc.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_

-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
@@ -46,9 +46,9 @@ typedef struct
    int16_t agcMode;
    AgcVad_t      vadNearend;
    AgcVad_t      vadFarend;
-#ifdef AGC_DEBUG
-    FILE*         logFile;
-    int           frameCounter;
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+    FILE* logFile;
+    int frameCounter;
 #endif
 } DigitalAgc_t;

--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -21,6 +21,7 @@
      'variables': {
        # Outputs some low-level debug files.
        'aec_debug_dump%': 0,
+        'agc_debug_dump%': 0,

        # Disables the usual mode where we trust the reported system delay
        # values the AEC receives. The corresponding define is set appropriately
@@ -93,6 +94,9 @@
        ['aec_untrusted_delay_for_testing==1', {
          'defines': ['WEBRTC_UNTRUSTED_DELAY',],
        }],
+        ['agc_debug_dump==1', {
+          'defines': ['WEBRTC_AGC_DEBUG_DUMP',],
+        }],
        ['enable_protobuf==1', {
          'dependencies': ['audioproc_debug_proto'],
          'defines': ['WEBRTC_AUDIOPROC_DEBUG_DUMP'],