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Merge :audio_processing and :aec_dump_interface.

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Merges the two targets in modules/audio_processing
and removes some redundant code. This enables not writing
a bunch of redundant code in
https://webrtc-review.googlesource.com/c/src/+/70502

':audio_processing' did depend on ':aec_dump_interface'.
'modules/audio_processing/aec_dump' did depend on
'aec_dump_interface' but not ':audio_processing'.

Having the AecDump implementation not depending on
'audio_processing' allows to have faster compilation time and
reduces the dependencies. However, maintaining such a decoupling
makes APM and AecDump client code more complex.

NOTRY=true # want this in and 'ios_api_framework' seems stuck.

Bug: webrtc:7404
Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c
Reviewed-on: https://webrtc-review.googlesource.com/76603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23244}
This commit is contained in:
Alex Loiko
2018-05-15 13:13:22 +02:00
committed by Commit Bot
parent 28a325b523
commit 1aec594018
8 changed files with 27 additions and 71 deletions
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22
modules/audio_processing/audio_processing_impl.cc
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@ -148,24 +148,6 @@ class HighPassFilterImpl : public HighPassFilter {
AudioProcessingImpl* apm_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(HighPassFilterImpl);
};
webrtc::InternalAPMStreamsConfig ToStreamsConfig(
const ProcessingConfig& api_format) {
webrtc::InternalAPMStreamsConfig result;
result.input_sample_rate = api_format.input_stream().sample_rate_hz();
result.input_num_channels = api_format.input_stream().num_channels();
result.output_num_channels = api_format.output_stream().num_channels();
result.render_input_num_channels =
api_format.reverse_input_stream().num_channels();
result.render_input_sample_rate =
api_format.reverse_input_stream().sample_rate_hz();
result.output_sample_rate = api_format.output_stream().sample_rate_hz();
result.render_output_sample_rate =
api_format.reverse_output_stream().sample_rate_hz();
result.render_output_num_channels =
api_format.reverse_output_stream().num_channels();
return result;
}
} // namespace
// Throughout webrtc, it's assumed that success is represented by zero.
@ -608,7 +590,7 @@ int AudioProcessingImpl::InitializeLocked() {
InitializePreProcessor();
if (aec_dump_) {
aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
aec_dump_->WriteInitMessage(formats_.api_format);
}
return kNoError;
}
@ -1638,7 +1620,7 @@ void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) {
// 'aec_dump' parameter, which is after locks are released.
aec_dump_.swap(aec_dump);
WriteAecDumpConfigMessage(true);
aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
aec_dump_->WriteInitMessage(formats_.api_format);
}
void AudioProcessingImpl::DetachAecDump() {