webrtc/modules/audio_processing/test
Steve Anton f254e9e9e5 Revert "Simplification and refactoring of the AudioBuffer code"
This reverts commit 81c0cf287c.

Reason for revert: internal test failures

Original change's description:
> Simplification and refactoring of the AudioBuffer code
> 
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
>  internal buffers in sync.
> -Removes all code relating to handling of fixed-point
>  sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
>  that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
>  more generic.
> 
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}

TBR=gustaf@webrtc.org,peah@webrtc.org

Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28931}
2019-08-21 18:00:59 +00:00
..
android/apmtest
conversational_speech Migrate WebRTC test infra to ABSL_FLAG. 2019-07-19 06:54:04 +00:00
py_quality_assessment Format almost everything. 2019-07-08 13:45:15 +00:00
aec_dump_based_simulator.cc audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
aec_dump_based_simulator.h audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
api_call_statistics.cc Added more refined benchmarking code for audioproc_f 2019-04-04 08:37:16 +00:00
api_call_statistics.h Added more refined benchmarking code for audioproc_f 2019-04-04 08:37:16 +00:00
apmtest.m
audio_buffer_tools.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_buffer_tools.h Format almost everything. 2019-07-08 13:45:15 +00:00
audio_processing_simulator.cc audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
audio_processing_simulator.h audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
audioproc_float_impl.cc audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
audioproc_float_impl.h audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
audioproc_float_main.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
bitexactness_tools.cc Format almost everything. 2019-07-08 13:45:15 +00:00
bitexactness_tools.h Add DCHECK and documentation to disallow trying to read more than two audio channels in helper function. 2018-01-12 15:27:50 +00:00
debug_dump_replayer.cc Add noise suppression settings to AudioProcessing::Config 2019-01-14 16:17:19 +00:00
debug_dump_replayer.h Store RuntimeSetting in Aec Dumps. 2018-09-10 11:40:28 +00:00
debug_dump_test.cc Fully qualify googletest symbols. 2019-04-09 17:18:20 +00:00
echo_canceller_test_tools.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_canceller_test_tools.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
echo_canceller_test_tools_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_control_mock.h APM unit test: echo path gain change events notified. 2019-01-10 11:06:24 +00:00
fake_recording_device.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
fake_recording_device.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fake_recording_device_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
performance_timer.cc Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
performance_timer.h Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
protobuf_utils.cc audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
protobuf_utils.h audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
runtime_setting_util.cc Add PlayoutVolumeChange RuntimeSetting. 2019-05-10 14:12:23 +00:00
runtime_setting_util.h Store RuntimeSetting in Aec Dumps. 2018-09-10 11:40:28 +00:00
simulator_buffers.cc Revert "Simplification and refactoring of the AudioBuffer code" 2019-08-21 18:00:59 +00:00
simulator_buffers.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
test_utils.cc audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
test_utils.h audioproc_f: input AEC dump as string, output audio to vector 2019-08-12 09:17:36 +00:00
unittest.proto Base ApmTest.Process on AudioProcessingStats.output_rms_dbfs 2018-12-18 16:45:03 +00:00
wav_based_simulator.cc Format almost everything. 2019-07-08 13:45:15 +00:00
wav_based_simulator.h Format almost everything. 2019-07-08 13:45:15 +00:00