webrtc/modules/audio_device/test/func_test_manager.cc - Issue 1172163004: Reformat existing code. There should be no functional effects.

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Unified Diff: webrtc/modules/audio_device/test/func_test_manager.cc

Issue 1172163004: Reformat existing code. There should be no functional effects. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Resync Created 5 years, 6 months ago

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Index: webrtc/modules/audio_device/test/func_test_manager.cc

diff --git a/webrtc/modules/audio_device/test/func_test_manager.cc b/webrtc/modules/audio_device/test/func_test_manager.cc

index efb7e7d9249e3e44f646f21a6e04927e1ac85208..ae3cd2c186db5714737c7e8542de0b20473cffe5 100644

--- a/webrtc/modules/audio_device/test/func_test_manager.cc

+++ b/webrtc/modules/audio_device/test/func_test_manager.cc

@@ -367,8 +367,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(

const uint16_t nSamplesIn = packet->nSamples;

const uint8_t nChannelsIn = packet->nChannels;

const uint32_t samplesPerSecIn = packet->samplesPerSec;

- const uint16_t nBytesPerSampleIn =

- packet->nBytesPerSample;

+ const uint16_t nBytesPerSampleIn = packet->nBytesPerSample;

int32_t fsInHz(samplesPerSecIn);

int32_t fsOutHz(samplesPerSec);

@@ -389,15 +388,14 @@ int32_t AudioTransportImpl::NeedMorePlayData(

{

_resampler.Push(

(const int16_t*) packet->dataBuffer,

- 2 * nSamplesIn,

- (int16_t*) audioSamples, 2

- * nSamples, lenOut);

+ 2 * nSamplesIn, (int16_t*) audioSamples,

+ 2 * nSamples, lenOut);

} else

{

_resampler.Push(

(const int16_t*) packet->dataBuffer,

- 2 * nSamplesIn, tmpBuf_96kHz, 2

- * nSamples, lenOut);

+ 2 * nSamplesIn, tmpBuf_96kHz, 2 * nSamples,

+ lenOut);

ptr16In = &tmpBuf_96kHz[0];

ptr16Out = (int16_t*) audioSamples;

@@ -429,16 +427,13 @@ int32_t AudioTransportImpl::NeedMorePlayData(

if (nChannels == 1)

{

_resampler.Push(

- (const int16_t*) packet->dataBuffer,

- nSamplesIn,

- (int16_t*) audioSamples,

- nSamples, lenOut);

+ (const int16_t*) packet->dataBuffer, nSamplesIn,

+ (int16_t*) audioSamples, nSamples, lenOut);

} else

{

_resampler.Push(

- (const int16_t*) packet->dataBuffer,

- nSamplesIn, tmpBuf_96kHz, nSamples,

- lenOut);

+ (const int16_t*) packet->dataBuffer, nSamplesIn,

+ tmpBuf_96kHz, nSamples, lenOut);

ptr16In = &tmpBuf_96kHz[0];

ptr16Out = (int16_t*) audioSamples;

@@ -472,8 +467,7 @@ int32_t AudioTransportImpl::NeedMorePlayData(

int16_t fileBuf[480];

// read mono-file

- int32_t len = _playFile.Read((int8_t*) fileBuf, 2

- * nSamples);

+ int32_t len = _playFile.Read((int8_t*) fileBuf, 2 * nSamples);

if (len != 2 * (int32_t) nSamples)

{

_playFile.Rewind();