Ensured that all files in APM are using the webrtc namespace

This CL adds namespaces to those files remaining within APM that do not
have any such.

BUG=webrtc:5298

Change-Id: I710b3d2a3644bea9d4bdffef0d77883b30303338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171111
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30850}

modules/audio_processing/aec_dump/aec_dump_unittest.cc

@@ -16,6 +16,8 @@
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"
 
+namespace webrtc {
+
 TEST(AecDumper, APICallsDoNotCrash) {
   // Note order of initialization: Task queue has to be initialized
   // before AecDump.
@@ -81,3 +83,5 @@ TEST(AecDumper, WriteToFile) {
   ASSERT_EQ(0, fclose(fid));
   ASSERT_EQ(0, remove(filename.c_str()));
 }
+
+}  // namespace webrtc

modules/audio_processing/agc/gain_map_internal.h

@@ -11,6 +11,8 @@
 #ifndef MODULES_AUDIO_PROCESSING_AGC_GAIN_MAP_INTERNAL_H_
 #define MODULES_AUDIO_PROCESSING_AGC_GAIN_MAP_INTERNAL_H_
 
+namespace webrtc {
+
 static const int kGainMapSize = 256;
 // Uses parameters: si = 2, sf = 0.25, D = 8/256
 static const int kGainMap[kGainMapSize] = {
@@ -33,4 +35,6 @@ static const int kGainMap[kGainMapSize] = {
     60,  60,  60,  61,  61,  61,  61,  62,  62,  62,  62,  63,  63,  63,  63,
     64};
 
+}  // namespace webrtc
+
 #endif  // MODULES_AUDIO_PROCESSING_AGC_GAIN_MAP_INTERNAL_H_

modules/audio_processing/agc/utility.cc

@@ -12,6 +12,8 @@
 
 #include <math.h>
 
+namespace webrtc {
+
 static const double kLog10 = 2.30258509299;
 static const double kLinear2DbScale = 20.0 / kLog10;
 static const double kLinear2LoudnessScale = 13.4 / kLog10;
@@ -33,3 +35,5 @@ double Db2Loudness(double db) {
 double Dbfs2Loudness(double dbfs) {
   return Db2Loudness(90 + dbfs);
 }
+
+}  // namespace webrtc

modules/audio_processing/agc/utility.h

@@ -11,6 +11,8 @@
 #ifndef MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_
 #define MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_
 
+namespace webrtc {
+
 // TODO(turajs): Add description of function.
 double Loudness2Db(double loudness);
 
@@ -20,4 +22,6 @@ double Db2Loudness(double db);
 
 double Dbfs2Loudness(double dbfs);
 
+}  // namespace webrtc
+
 #endif  // MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_

modules/audio_processing/transient/windows_private.h

@@ -11,6 +11,8 @@
 #ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_WINDOWS_PRIVATE_H_
 #define MODULES_AUDIO_PROCESSING_TRANSIENT_WINDOWS_PRIVATE_H_
 
+namespace webrtc {
+
 // Hanning window for 4ms 16kHz
 static const float kHanning64w128[128] = {
     0.00000000000000f, 0.02454122852291f, 0.04906767432742f, 0.07356456359967f,
@@ -550,4 +552,6 @@ static const float kBlocks480w1024[1024] = {
     0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
     0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f};
 
+}  // namespace webrtc
+
 #endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_WINDOWS_PRIVATE_H_

modules/audio_processing/utility/delay_estimator.cc

@@ -17,6 +17,10 @@
 
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
+namespace {
+
 // Number of right shifts for scaling is linearly depending on number of bits in
 // the far-end binary spectrum.
 static const int kShiftsAtZero = 13;  // Right shifts at zero binary spectrum.
@@ -38,6 +42,8 @@ static const float kFractionSlope = 0.05f;
 static const float kMinFractionWhenPossiblyCausal = 0.5f;
 static const float kMinFractionWhenPossiblyNonCausal = 0.25f;
 
+}  // namespace
+
 // Counts and returns number of bits of a 32-bit word.
 static int BitCount(uint32_t u32) {
   uint32_t tmp =
@@ -698,3 +704,5 @@ void WebRtc_MeanEstimatorFix(int32_t new_value,
   }
   *mean_value += diff;
 }
+
+}  // namespace webrtc

modules/audio_processing/utility/delay_estimator.h

@@ -16,6 +16,8 @@
 
 #include <stdint.h>
 
+namespace webrtc {
+
 static const int32_t kMaxBitCountsQ9 = (32 << 9);  // 32 matching bits in Q9.
 
 typedef struct {
@@ -250,4 +252,6 @@ void WebRtc_MeanEstimatorFix(int32_t new_value,
                              int factor,
                              int32_t* mean_value);
 
+}  // namespace webrtc
+
 #endif  // MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_

modules/audio_processing/utility/delay_estimator_internal.h

@@ -15,6 +15,8 @@
 
 #include "modules/audio_processing/utility/delay_estimator.h"
 
+namespace webrtc {
+
 typedef union {
   float float_;
   int32_t int32_;
@@ -44,4 +46,6 @@ typedef struct {
   BinaryDelayEstimator* binary_handle;
 } DelayEstimator;
 
+}  // namespace webrtc
+
 #endif  // MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_INTERNAL_H_

modules/audio_processing/utility/delay_estimator_unittest.cc

@@ -14,6 +14,8 @@
 #include "modules/audio_processing/utility/delay_estimator_wrapper.h"
 #include "test/gtest.h"
 
+namespace webrtc {
+
 namespace {
 
 enum { kSpectrumSize = 65 };
@@ -615,3 +617,5 @@ TEST_F(DelayEstimatorTest, VerifyHistorySizeIsSetAndKeptAfterInit) {
 // TODO(bjornv): Add tests for SoftReset...(...).
 
 }  // namespace
+
+}  // namespace webrtc

modules/audio_processing/utility/delay_estimator_wrapper.cc

@@ -17,6 +17,8 @@
 #include "modules/audio_processing/utility/delay_estimator_internal.h"
 #include "rtc_base/checks.h"
 
+namespace webrtc {
+
 // Only bit |kBandFirst| through bit |kBandLast| are processed and
 // |kBandFirst| - |kBandLast| must be < 32.
 enum { kBandFirst = 12 };
@@ -483,3 +485,5 @@ float WebRtc_last_delay_quality(void* handle) {
   RTC_DCHECK(self);
   return WebRtc_binary_last_delay_quality(self->binary_handle);
 }
+
+}  // namespace webrtc

modules/audio_processing/utility/delay_estimator_wrapper.h

@@ -16,6 +16,8 @@
 
 #include <stdint.h>
 
+namespace webrtc {
+
 // Releases the memory allocated by WebRtc_CreateDelayEstimatorFarend(...)
 void WebRtc_FreeDelayEstimatorFarend(void* handle);
 
@@ -241,4 +243,6 @@ int WebRtc_last_delay(void* handle);
 //      - delay_quality : >= 0  - Estimation quality of last calculated delay.
 float WebRtc_last_delay_quality(void* handle);
 
+}  // namespace webrtc
+
 #endif  // MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_

modules/audio_processing/vad/noise_gmm_tables.h

@@ -13,6 +13,8 @@
 #ifndef MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
 #define MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
 
+namespace webrtc {
+
 static const int kNoiseGmmNumMixtures = 12;
 static const int kNoiseGmmDim = 3;
 
@@ -74,4 +76,7 @@ static const double kNoiseGmmWeights[kNoiseGmmNumMixtures] = {
     -1.79789356118641e+01, -1.42830169160894e+01, -1.56500228061379e+01,
     -1.83124990950113e+01, -1.69979436177477e+01, -1.12329424387828e+01,
     -1.41311785780639e+01, -1.47171861448585e+01, -1.35963362781839e+01};
+
+}  // namespace webrtc
+
 #endif  // MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_

modules/audio_processing/vad/pitch_based_vad.h

@@ -53,4 +53,5 @@ class PitchBasedVad {
 };
 
 }  // namespace webrtc
+
 #endif  // MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_

modules/audio_processing/vad/pitch_internal.cc

@@ -12,6 +12,8 @@
 
 #include <cmath>
 
+namespace webrtc {
+
 // A 4-to-3 linear interpolation.
 // The interpolation constants are derived as following:
 // Input pitch parameters are updated every 7.5 ms. Within a 30-ms interval
@@ -49,3 +51,5 @@ void GetSubframesPitchParameters(int sampling_rate_hz,
     pitch_lag_hz[n] = (sampling_rate_hz) / (pitch_lag_hz[n]);
   }
 }
+
+}  // namespace webrtc

modules/audio_processing/vad/pitch_internal.h

@@ -11,6 +11,8 @@
 #ifndef MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
 #define MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
 
+namespace webrtc {
+
 // TODO(turajs): Write a description of this function. Also be consistent with
 // usage of |sampling_rate_hz| vs |kSamplingFreqHz|.
 void GetSubframesPitchParameters(int sampling_rate_hz,
@@ -23,4 +25,6 @@ void GetSubframesPitchParameters(int sampling_rate_hz,
                                  double* log_pitch_gain,
                                  double* pitch_lag_hz);
 
+}  // namespace webrtc
+
 #endif  // MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_

modules/audio_processing/vad/pitch_internal_unittest.cc

@@ -14,6 +14,8 @@
 
 #include "test/gtest.h"
 
+namespace webrtc {
+
 TEST(PitchInternalTest, test) {
   const int kSamplingRateHz = 8000;
   const int kNumInputParameters = 4;
@@ -48,3 +50,5 @@ TEST(PitchInternalTest, test) {
   EXPECT_NEAR(old_lag, expected_old_lag, 1e-6);
   EXPECT_NEAR(log_old_gain, expected_log_old_gain, 1e-8);
 }
+
+}  // namespace webrtc