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platform-external-webrtc/webrtc/modules/audio_coding/neteq
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henrik.lundin@webrtc.org c8e98187d1 Receiver bit-exactness test for AudioCoding Module 
This CL introduces a bit-exactness test for the receive-side of the
AudioCoding Module. The main part of the test is done in the helper
class AcmReceiveTest. The test is executed from the test fixture
AcmReceiverBitExactness.

The test inserts packets from a pre-encoded RTP file. The output is
summed up into a checksum, which is verified versus a reference at the
end of the test. Alternatively, if the flag --generate_output is given,
the output is written to a file for subjective verification.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-26 19:07:04 +00:00
..
interface
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mock
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test
This is to compare NetEq with various codecs under a shared packet loss pattern.
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tools
Receiver bit-exactness test for AudioCoding Module
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accelerate.cc
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accelerate.h
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audio_classifier_unittest.cc
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audio_classifier.cc
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audio_classifier.h
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audio_decoder_impl.cc
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audio_decoder_impl.h
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audio_decoder_unittest.cc
Remove payload duplication in AudioDecoderTest
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audio_decoder_unittests.isolate
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audio_decoder.cc
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
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audio_multi_vector_unittest.cc
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audio_multi_vector.cc
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audio_multi_vector.h
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audio_vector_unittest.cc
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audio_vector.cc
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audio_vector.h
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background_noise_unittest.cc
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background_noise.cc
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background_noise.h
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buffer_level_filter_unittest.cc
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buffer_level_filter.cc
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buffer_level_filter.h
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comfort_noise_unittest.cc
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comfort_noise.cc
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comfort_noise.h
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decision_logic_fax.cc
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decision_logic_fax.h
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decision_logic_normal.cc
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decision_logic_normal.h
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decision_logic_unittest.cc
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decision_logic.cc
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decision_logic.h
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decoder_database_unittest.cc
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decoder_database.cc
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decoder_database.h
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defines.h
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delay_manager_unittest.cc
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delay_manager.cc
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delay_manager.h
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delay_peak_detector_unittest.cc
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delay_peak_detector.cc
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delay_peak_detector.h
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dsp_helper_unittest.cc
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dsp_helper.cc
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dsp_helper.h
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dtmf_buffer_unittest.cc
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dtmf_buffer.cc
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dtmf_buffer.h
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dtmf_tone_generator_unittest.cc
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dtmf_tone_generator.cc
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dtmf_tone_generator.h
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expand_unittest.cc
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expand.cc
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expand.h
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merge_unittest.cc
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merge.cc
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merge.h
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neteq_external_decoder_unittest.cc
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neteq_impl_unittest.cc
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neteq_impl.cc
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neteq_impl.h
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neteq_stereo_unittest.cc
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neteq_tests.gypi
This is to compare NetEq with various codecs under a shared packet loss pattern.
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neteq_unittest.cc
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neteq.cc
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neteq.gypi
Adding test::AudioSink interface and derived classes
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normal_unittest.cc
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normal.cc
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normal.h
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OWNERS
Add kjellander@webrtc.org as OWNER for *.isolate
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packet_buffer_unittest.cc
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packet_buffer.cc
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packet_buffer.h
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packet.h
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payload_splitter_unittest.cc
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
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payload_splitter.cc
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payload_splitter.h
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post_decode_vad_unittest.cc
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post_decode_vad.cc
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post_decode_vad.h
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preemptive_expand.cc
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preemptive_expand.h
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random_vector_unittest.cc
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random_vector.cc
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random_vector.h
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rtcp.cc
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rtcp.h
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statistics_calculator.cc
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statistics_calculator.h
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sync_buffer_unittest.cc
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sync_buffer.cc
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sync_buffer.h
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time_stretch_unittest.cc
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time_stretch.cc
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time_stretch.h
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timestamp_scaler_unittest.cc
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
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timestamp_scaler.cc
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
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timestamp_scaler.h
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