Formatting ACM tests
Pure formatting of all files located in /webrtc/modules/audio_coding/main/test/ Smaller manual modifications done after using Eclipse formatting tool, like wrapping long lines (mostly comments). BUG=issue1024 Review URL: https://webrtc-codereview.appspot.com/1342004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3946 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -78,8 +78,7 @@ int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type,
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rtp_info.type.Audio.channel = 1;
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memcpy(payload_data_, payload_data, payload_size);
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status = receiver_acm_->IncomingPacket(payload_data_, payload_size,
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rtp_info);
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status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);
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payload_size_ = payload_size;
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timestamp_diff_ = timestamp - last_in_timestamp_;
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@ -127,8 +126,8 @@ TestAllCodecs::~TestAllCodecs() {
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}
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void TestAllCodecs::Perform() {
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const std::string file_name =
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webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
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const std::string file_name = webrtc::test::ResourcePath(
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"audio_coding/testfile32kHz", "pcm");
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infile_a_.Open(file_name, 32000, "rb");
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if (test_mode_ == 0) {
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@ -725,9 +724,9 @@ void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
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// packet. If variable rate codec (extra_byte == -1), set to -1 (65535).
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if (extra_byte != -1) {
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// Add 0.875 to always round up to a whole byte
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packet_size_bytes_ =
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static_cast<uint16_t>(static_cast<float>(packet_size * rate) /
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static_cast<float>(sampling_freq_hz * 8) + 0.875) + extra_byte;
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packet_size_bytes_ = static_cast<uint16_t>(static_cast<float>(packet_size
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* rate) / static_cast<float>(sampling_freq_hz * 8) + 0.875)
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+ extra_byte;
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} else {
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// Packets will have a variable size.
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packet_size_bytes_ = -1;
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