Cast: Refactor Audio Receiver to Clang format

media/cast/audio_receiver/audio_decoder_unittest.cc

Index: media/cast/audio_receiver/audio_decoder_unittest.cc
diff --git a/media/cast/audio_receiver/audio_decoder_unittest.cc b/media/cast/audio_receiver/audio_decoder_unittest.cc
index f01f0efac1bb28bf9155272a7e2c68240f8d8f1d..962217d9f2b462022d938a83e16eb9bd8e0451c5 100644
--- a/media/cast/audio_receiver/audio_decoder_unittest.cc
+++ b/media/cast/audio_receiver/audio_decoder_unittest.cc
@@ -30,10 +30,15 @@ class AudioDecoderTest : public ::testing::Test {
     testing_clock_ = new base::SimpleTestTickClock();
     testing_clock_->Advance(base::TimeDelta::FromMilliseconds(1234));
     task_runner_ = new test::FakeTaskRunner(testing_clock_);
-    cast_environment_ = new CastEnvironment(
-        scoped_ptr<base::TickClock>(testing_clock_).Pass(), task_runner_,
-        task_runner_, task_runner_, task_runner_, task_runner_, task_runner_,
-        GetDefaultCastReceiverLoggingConfig());
+    cast_environment_ =
+        new CastEnvironment(scoped_ptr<base::TickClock>(testing_clock_).Pass(),
+                            task_runner_,
+                            task_runner_,
+                            task_runner_,
+                            task_runner_,
+                            task_runner_,
+                            task_runner_,
+                            GetDefaultCastReceiverLoggingConfig());
   }
   virtual ~AudioDecoderTest() {}
 
@@ -48,6 +53,8 @@ class AudioDecoderTest : public ::testing::Test {
   scoped_refptr<test::FakeTaskRunner> task_runner_;
   scoped_refptr<CastEnvironment> cast_environment_;
   scoped_ptr<AudioDecoder> audio_decoder_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioDecoderTest);
 };
 
 TEST_F(AudioDecoderTest, Pcm16MonoNoResampleOnePacket) {
@@ -75,21 +82,17 @@ TEST_F(AudioDecoderTest, Pcm16MonoNoResampleOnePacket) {
   PcmAudioFrame audio_frame;
   uint32 rtp_timestamp;
 
-  EXPECT_FALSE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
-                                                desired_frequency,
-                                                &audio_frame,
-                                                &rtp_timestamp));
+  EXPECT_FALSE(audio_decoder_->GetRawAudioFrame(

[hubbe, 2014/01/29 20:07:39]

Hmm, this isn't actually easier to read IMHO.
Does the tool really complain about the way it was indented before?

[mikhal1, 2014/01/29 21:02:38]

Apparently the tool prefers everything in one line when possible. Would rather
stick with the tool for future ease of use. Once we decide on a format, I think
we need to stick with it.
On 2014/01/29 20:07:39, hubbe wrote:

+      number_of_10ms_blocks, desired_frequency, &audio_frame, &rtp_timestamp));
 
   uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
   size_t payload_size = payload.size() * sizeof(int16);
 
-  audio_decoder_->IncomingParsedRtpPacket(payload_data,
-      payload_size, rtp_header);
+  audio_decoder_->IncomingParsedRtpPacket(
+      payload_data, payload_size, rtp_header);
 
-  EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
-                                               desired_frequency,
-                                               &audio_frame,
-                                               &rtp_timestamp));
+  EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(
+      number_of_10ms_blocks, desired_frequency, &audio_frame, &rtp_timestamp));
   EXPECT_EQ(1, audio_frame.channels);
   EXPECT_EQ(16000, audio_frame.frequency);
   EXPECT_EQ(640ul, audio_frame.samples.size());
@@ -125,24 +128,22 @@ TEST_F(AudioDecoderTest, Pcm16StereoNoResampleTwoPackets) {
   uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
   size_t payload_size = payload.size() * sizeof(int16);
 
-  audio_decoder_->IncomingParsedRtpPacket(payload_data,
-      payload_size, rtp_header);
+  audio_decoder_->IncomingParsedRtpPacket(
+      payload_data, payload_size, rtp_header);
 
   int number_of_10ms_blocks = 2;
   int desired_frequency = 16000;
   PcmAudioFrame audio_frame;
   uint32 rtp_timestamp;
 
-  EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
-                                               desired_frequency,
-                                               &audio_frame,
-                                               &rtp_timestamp));
+  EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(
+      number_of_10ms_blocks, desired_frequency, &audio_frame, &rtp_timestamp));
   EXPECT_EQ(2, audio_frame.channels);
   EXPECT_EQ(16000, audio_frame.frequency);
   EXPECT_EQ(640ul, audio_frame.samples.size());
   // First 10 samples per channel are 0 from NetEq.
   for (size_t i = 10 * audio_config.channels; i < audio_frame.samples.size();
-      ++i) {
+       ++i) {
     EXPECT_EQ(0x3412, audio_frame.samples[i]);
   }
 
@@ -150,13 +151,11 @@ TEST_F(AudioDecoderTest, Pcm16StereoNoResampleTwoPackets) {
   rtp_header.webrtc.header.sequenceNumber++;
   rtp_header.webrtc.header.timestamp += (audio_config.frequency / 100) * 2 * 2;
 
-  audio_decoder_->IncomingParsedRtpPacket(payload_data,
-      payload_size, rtp_header);
+  audio_decoder_->IncomingParsedRtpPacket(
+      payload_data, payload_size, rtp_header);
 
-  EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
-                                               desired_frequency,
-                                               &audio_frame,
-                                               &rtp_timestamp));
+  EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(
+      number_of_10ms_blocks, desired_frequency, &audio_frame, &rtp_timestamp));
   EXPECT_EQ(2, audio_frame.channels);
   EXPECT_EQ(16000, audio_frame.frequency);
   EXPECT_EQ(640ul, audio_frame.samples.size());
@@ -194,18 +193,16 @@ TEST_F(AudioDecoderTest, Pcm16Resample) {
   uint8* payload_data = reinterpret_cast<uint8*>(&payload[0]);
   size_t payload_size = payload.size() * sizeof(int16);
 
-  audio_decoder_->IncomingParsedRtpPacket(payload_data,
-      payload_size, rtp_header);
+  audio_decoder_->IncomingParsedRtpPacket(
+      payload_data, payload_size, rtp_header);
 
   int number_of_10ms_blocks = 2;
   int desired_frequency = 48000;
   PcmAudioFrame audio_frame;
   uint32 rtp_timestamp;
 
-  EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(number_of_10ms_blocks,
-                                               desired_frequency,
-                                               &audio_frame,
-                                               &rtp_timestamp));
+  EXPECT_TRUE(audio_decoder_->GetRawAudioFrame(
+      number_of_10ms_blocks, desired_frequency, &audio_frame, &rtp_timestamp));
 
   EXPECT_EQ(2, audio_frame.channels);
   EXPECT_EQ(48000, audio_frame.frequency);
@@ -213,9 +210,10 @@ TEST_F(AudioDecoderTest, Pcm16Resample) {
   int count = 0;
   // Resampling makes the variance worse.
   for (size_t i = 100 * audio_config.channels; i < audio_frame.samples.size();
-      ++i) {
+       ++i) {
     EXPECT_NEAR(0x3412, audio_frame.samples[i], 400);
-    if (0x3412 == audio_frame.samples[i])  count++;
+    if (0x3412 == audio_frame.samples[i])
+      count++;
   }
 }