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Fixing lint errors in NetEq4

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Just taking care of a few old lint errors.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5359 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2014-01-09 14:01:55 +00:00
parent c5aeb2aa15
commit e7ce437333
6 changed files with 13 additions and 14 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
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@@ -74,8 +74,7 @@ void AudioMultiVector::PushBackInterleaved(const int16_t* append_this,
return; return;
} }
size_t length_per_channel = length / num_channels_; size_t length_per_channel = length / num_channels_;
int16_t* temp_array = int16_t* temp_array = new int16_t[length_per_channel]; // Temporary storage.
new int16_t[length_per_channel]; // Intermediate storage.
for (size_t channel = 0; channel < num_channels_; ++channel) { for (size_t channel = 0; channel < num_channels_; ++channel) {
// Copy elements to |temp_array|. // Copy elements to |temp_array|.
// Set |source_ptr| to first element of this channel. // Set |source_ptr| to first element of this channel.

12
webrtc/modules/audio_coding/neteq4/neteq_impl.cc
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@@ -143,12 +143,12 @@ int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
int error = InsertPacketInternal( int error = InsertPacketInternal(
rtp_header, kSyncPayload, sizeof(kSyncPayload), receive_timestamp, true); rtp_header, kSyncPayload, sizeof(kSyncPayload), receive_timestamp, true);
if (error != 0) { if (error != 0) {
LOG_FERR1(LS_WARNING, InsertPacketInternal, error); LOG_FERR1(LS_WARNING, InsertPacketInternal, error);
error_code_ = error; error_code_ = error;
return kFail; return kFail;
} }
return kOK; return kOK;
} }
int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio, int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,

5
webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
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@@ -999,7 +999,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
// Even if there is RTP packet in NetEq's buffer, the first frame pulled // Even if there is RTP packet in NetEq's buffer, the first frame pulled
// from NetEq starts with few zero samples. Here we measure this delay. // from NetEq starts with few zero samples. Here we measure this delay.
if (n == 0) { if (n == 0) {
while(decoded[delay_samples] == 0) delay_samples++; while (decoded[delay_samples] == 0) delay_samples++;
} }
rtp_info.header.sequenceNumber++; rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz; rtp_info.header.timestamp += kBlockSize16kHz;
@@ -1182,7 +1182,6 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
// Expect delay (in samples) to be less than 2 packets. // Expect delay (in samples) to be less than 2 packets.
EXPECT_LE(timestamp - neteq_->PlayoutTimestamp(), EXPECT_LE(timestamp - neteq_->PlayoutTimestamp(),
static_cast<uint32_t>(kSamples * 2)); static_cast<uint32_t>(kSamples * 2));
} }
// Make sure we have actually tested wrap-around. // Make sure we have actually tested wrap-around.
ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped); ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
@@ -1216,4 +1215,4 @@ TEST_F(NetEqDecodingTest, TimestampAndSequenceNumberWrap) {
WrapTest(0xFFFF - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true); WrapTest(0xFFFF - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true);
} }
} // namespace } // namespace webrtc

2
webrtc/modules/audio_coding/neteq4/normal.h
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@@ -65,4 +65,4 @@ class Normal {
}; };
} // namespace webrtc } // namespace webrtc
#endif // SRC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_ #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_

2
webrtc/modules/audio_coding/neteq4/random_vector.cc
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@@ -54,4 +54,4 @@ void RandomVector::IncreaseSeedIncrement(int16_t increase_by) {
seed_increment_+= increase_by; seed_increment_+= increase_by;
seed_increment_ &= kRandomTableSize - 1; seed_increment_ &= kRandomTableSize - 1;
} }
} } // namespace webrtc

3
webrtc/modules/audio_coding/neteq4/rtcp.cc
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@@ -10,9 +10,10 @@
#include "webrtc/modules/audio_coding/neteq4/rtcp.h" #include "webrtc/modules/audio_coding/neteq4/rtcp.h"
#include <algorithm>
#include <string.h> #include <string.h>
#include <algorithm>
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/interface/module_common_types.h" #include "webrtc/modules/interface/module_common_types.h"