Cast: Refactor Audio Receiver to Clang format

media/cast/audio_receiver/audio_decoder_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/audio_receiver/audio_decoder.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/test/fake_task_runner.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
@@ skipping 12 matching lines @@
   }
 };
 }  // namespace.
 
 class AudioDecoderTest : public ::testing::Test {
  protected:
   AudioDecoderTest() {
     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() {}
 
   void Configure(const AudioReceiverConfig& audio_config) {
     audio_decoder_.reset(
         new AudioDecoder(cast_environment_, audio_config, &cast_feedback_));
   }
 
   TestRtpPayloadFeedback cast_feedback_;
   // Owned by CastEnvironment.
   base::SimpleTestTickClock* testing_clock_;
   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) {
   AudioReceiverConfig audio_config;
   audio_config.rtp_payload_type = 127;
   audio_config.frequency = 16000;
   audio_config.channels = 1;
   audio_config.codec = transport::kPcm16;
   audio_config.use_external_decoder = false;
   Configure(audio_config);
 
   RtpCastHeader rtp_header;
   rtp_header.webrtc.header.payloadType = 127;
   rtp_header.webrtc.header.sequenceNumber = 1234;
   rtp_header.webrtc.header.timestamp = 0x87654321;
   rtp_header.webrtc.header.ssrc = 0x12345678;
   rtp_header.webrtc.header.paddingLength = 0;
   rtp_header.webrtc.header.headerLength = 12;
   rtp_header.webrtc.type.Audio.channel = 1;
   rtp_header.webrtc.type.Audio.isCNG = false;
 
   std::vector<int16> payload(640, 0x1234);
   int number_of_10ms_blocks = 4;
   int desired_frequency = 16000;
   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());
   // First 10 samples per channel are 0 from NetEq.
   for (size_t i = 10; i < audio_frame.samples.size(); ++i) {
     EXPECT_EQ(0x3412, audio_frame.samples[i]);
   }
 }
 
 TEST_F(AudioDecoderTest, Pcm16StereoNoResampleTwoPackets) {
@@ skipping 15 matching lines @@
   rtp_header.webrtc.header.headerLength = 12;
 
   rtp_header.webrtc.type.Audio.isCNG = false;
   rtp_header.webrtc.type.Audio.channel = 2;
 
   std::vector<int16> payload(640, 0x1234);
 
   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]);
   }
 
   rtp_header.frame_id++;
   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());
   for (size_t i = 0; i < audio_frame.samples.size(); ++i) {
     EXPECT_NEAR(0x3412, audio_frame.samples[i], 1000);
   }
   // Test cast callback.
   audio_decoder_->SendCastMessage();
   testing_clock_->Advance(base::TimeDelta::FromMilliseconds(33));
   audio_decoder_->SendCastMessage();
@@ skipping 17 matching lines @@
   rtp_header.webrtc.header.headerLength = 12;
 
   rtp_header.webrtc.type.Audio.isCNG = false;
   rtp_header.webrtc.type.Audio.channel = 2;
 
   std::vector<int16> payload(640, 0x1234);
 
   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);
   EXPECT_EQ(1920ul, audio_frame.samples.size());  // Upsampled to 48 KHz.
   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++;
   }
 }
 
 }  // namespace cast
 }  // namespace media