These 14 commits are when the Protocol Buffers files have changed:
| Commit: | 4f1658c | |
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| Author: | andrew@webrtc.org | |
Support arbitrary input/output rates and downmixing in AudioProcessing. Select "processing" rates based on the input and output sampling rates. Resample the input streams to those rates, and if necessary to the output rate. - Remove deprecated stream format APIs. - Remove deprecated device sample rate APIs. - Add a ChannelBuffer class to help manage deinterleaved channels. - Clean up the splitting filter state. - Add a unit test which verifies the output against known-working native format output. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
The documentation is generated from this commit.
| Commit: | 0ee472a | |
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| Author: | andrew@webrtc.org | |
Add tests and modify tools for new float deinterleaved interface. - Add an Initialize() overload to allow specification of format parameters. This is mainly useful for testing, but could be used in the cases where a consumer knows the format before the streams arrive. - Add a reverse_sample_rate_hz_ parameter to prepare for mismatched capture and render rates. There is no functional change as it is currently constrained to match the capture rate. - Fix a bug in the float dump: we need to use add_ rather than set_. - Add a debug dump test for both int and float interfaces. - Enable unpacking of float dumps. - Enable audioproc to read float dumps. - Move more shared functionality to test_utils.h, and generally tidy up a bit by consolidating repeated code. BUG=2894 TESTED=Verified that the output produced by the float debug dump test is correct. Processed the resulting debug dump file with audioproc and ensured that we get identical output. (This is crucial, as we need to be able to exactly reproduce online results offline.) R=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5676 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 8ba8ab6 | |
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| Author: | andrew@webrtc.org | |
Add a deinterleaved float interface to AudioProcessing. This is mainly to support the native audio format in Chrome. Although this implementation just moves the float->int conversion under the hood, we will transition AudioProcessing towards supporting this format throughout. - Add a test which verifies we get identical output with the float and int interfaces. - The float and int wrappers are tasked with conversion to the AudioBuffer format. A new shared Process/Analyze method does most of the work. - Add a new field to the debug.proto to hold deinterleaved data. - Add helpers to audio_utils.cc, and start using numeric_limits. - Note that there was no performance difference between numeric_limits and a literal value when measured on Linux using gcc or clang. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, henrikg@webrtc.org, tommi@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5641 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 3c19592 | |
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| Author: | andrew@webrtc.org | |
Add a keypress field to the audioproc debug proto. Log the value in AudioProcessing, and unpack it to a new file in the unpacking tool. TESTED= - The new tool can unpack old dumps. - The old tool can unpack new dumps (without keypress.bool). - Unpacking a new dump from voe_cmd_test produces a keypress.bool that appears correct when examined. R=aluebs@webrtc.org, bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8509005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5535 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 14b43be | |
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| Author: | andrew@webrtc.org | |
Move src/ -> webrtc/ TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 08329f4 | |
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| Author: | bjornv@webrtc.org | |
Added API to port internal speech probability in NS. Identical with CL652007 that's already been accepted for commit. TBR=andrew@webrtc.org BUG=None TEST=None Review URL: https://webrtc-codereview.appspot.com/670009 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2511 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 755b04a | |
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| Author: | andrew@webrtc.org | |
Add RMS computation for the RTP level indicator. - Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer. - We now use the entire packet rather than the last 10 ms frame. - Restore functionality to LevelEstimator. - Fix a bug in the splitting filter. - Fix a number of bugs in process_test related to a poorly named AudioFrame member. - Update the unittest protobuf and float reference output. - Add audioproc unittests. - Reenable voe_extended_tests, and add a real function test. - Use correct minimum level of 127. TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test Review URL: http://webrtc-codereview.appspot.com/279003 git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 4d5d5c1 | |
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| Author: | andrew@webrtc.org | |
Reorganize the audio_processing source. - Remove main and source directories. - Change .gyp, .gypi and Android.mk files correspondingly. No other source changes. Review URL: http://webrtc-codereview.appspot.com/241001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@767 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 1ba3dbe | |
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| Author: | bjornv@google.com | |
Adds possibility to log delay estimates in AEC. Review URL: http://webrtc-codereview.appspot.com/178001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@674 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 808e0e0 | |
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| Author: | ajm@google.com | |
Update the debug recordings to use protobufs. Also modify the unittest proto based to correspond with the changes. process_test is a bit of a hack job, but it works fine and isn't too unreadable. We should refactor it properly later. Review URL: http://webrtc-codereview.appspot.com/98007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@296 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | a769fa5 | |
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| Author: | ajm@google.com | |
Adding more output data checks to APM unittest. Blowing out the protobuf definition (changing the tags) since we're still in the formative stages. Later, this would be very bad. Leaving a Frame message in case we want frame-by-frame data, but we prefer to keep the output storage small in general so avoiding it thus far. Review URL: http://webrtc-codereview.appspot.com/68004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@203 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 5adc73a | |
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| Author: | niklase@google.com | |
git-svn-id: http://webrtc.googlecode.com/svn/trunk@166 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 470e71d | |
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| Author: | niklase@google.com | |
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
| Commit: | 7c4469b | |
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| Author: | ajm@google.com | |
Revamp of audio_processing unit test to use protocol buffers. Chromium's protobuf version is synced to third_party. This isn't really needed for the unit test, but I'd like to use it soon for echo recordings, so I used this as a warm up. Review URL: http://webrtc-codereview.appspot.com/56002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@151 4adac7df-926f-26a2-2b94-8c16560cd09d