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Add keyboard channel support to AudioBuffer.

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Also use local aliases for AudioBuffers for brevity.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5973 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2014-04-24 18:28:56 +00:00
parent d57b8149c2
commit 103657b484
4 changed files with 135 additions and 41 deletions
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70
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -468,48 +468,46 @@ int AudioProcessingImpl::ProcessStreamLocked() {
}
#endif
AudioBuffer* ca = capture_audio_.get(); // For brevity.
bool data_processed = is_data_processed();
if (analysis_needed(data_processed)) {
for (int i = 0; i < fwd_proc_format_.num_channels(); i++) {
SplitFilterStates* filter_states = capture_audio_->filter_states(i);
// Split into a low and high band.
WebRtcSpl_AnalysisQMF(capture_audio_->data(i),
capture_audio_->samples_per_channel(),
capture_audio_->low_pass_split_data(i),
capture_audio_->high_pass_split_data(i),
filter_states->analysis_filter_state1,
filter_states->analysis_filter_state2);
WebRtcSpl_AnalysisQMF(ca->data(i),
ca->samples_per_channel(),
ca->low_pass_split_data(i),
ca->high_pass_split_data(i),
ca->filter_states(i)->analysis_filter_state1,
ca->filter_states(i)->analysis_filter_state2);
}
}
RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(capture_audio_.get()));
RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(capture_audio_.get()));
RETURN_ON_ERR(echo_cancellation_->ProcessCaptureAudio(capture_audio_.get()));
RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(ca));
RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(ca));
RETURN_ON_ERR(echo_cancellation_->ProcessCaptureAudio(ca));
if (echo_control_mobile_->is_enabled() && noise_suppression_->is_enabled()) {
capture_audio_->CopyLowPassToReference();
ca->CopyLowPassToReference();
}
RETURN_ON_ERR(noise_suppression_->ProcessCaptureAudio(capture_audio_.get()));
RETURN_ON_ERR(
echo_control_mobile_->ProcessCaptureAudio(capture_audio_.get()));
RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(capture_audio_.get()));
RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(capture_audio_.get()));
RETURN_ON_ERR(noise_suppression_->ProcessCaptureAudio(ca));
RETURN_ON_ERR(echo_control_mobile_->ProcessCaptureAudio(ca));
RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(ca));
RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(ca));
if (synthesis_needed(data_processed)) {
for (int i = 0; i < fwd_proc_format_.num_channels(); i++) {
// Recombine low and high bands.
SplitFilterStates* filter_states = capture_audio_->filter_states(i);
WebRtcSpl_SynthesisQMF(capture_audio_->low_pass_split_data(i),
capture_audio_->high_pass_split_data(i),
capture_audio_->samples_per_split_channel(),
capture_audio_->data(i),
filter_states->synthesis_filter_state1,
filter_states->synthesis_filter_state2);
WebRtcSpl_SynthesisQMF(ca->low_pass_split_data(i),
ca->high_pass_split_data(i),
ca->samples_per_split_channel(),
ca->data(i),
ca->filter_states(i)->synthesis_filter_state1,
ca->filter_states(i)->synthesis_filter_state2);
}
}
// The level estimator operates on the recombined data.
RETURN_ON_ERR(level_estimator_->ProcessStream(capture_audio_.get()));
RETURN_ON_ERR(level_estimator_->ProcessStream(ca));
was_stream_delay_set_ = false;
return kNoError;
@ -592,27 +590,23 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
return AnalyzeReverseStreamLocked();
}
// TODO(ajm): Have AnalyzeReverseStream accept sample rates not matching the
// primary stream and convert ourselves rather than having the user manage it.
// We can be smarter and use the splitting filter when appropriate. Similarly,
// perform downmixing here.
int AudioProcessingImpl::AnalyzeReverseStreamLocked() {
AudioBuffer* ra = render_audio_.get(); // For brevity.
if (rev_proc_format_.rate() == kSampleRate32kHz) {
for (int i = 0; i < rev_proc_format_.num_channels(); i++) {
// Split into low and high band.
SplitFilterStates* filter_states = render_audio_->filter_states(i);
WebRtcSpl_AnalysisQMF(render_audio_->data(i),
render_audio_->samples_per_channel(),
render_audio_->low_pass_split_data(i),
render_audio_->high_pass_split_data(i),
filter_states->analysis_filter_state1,
filter_states->analysis_filter_state2);
WebRtcSpl_AnalysisQMF(ra->data(i),
ra->samples_per_channel(),
ra->low_pass_split_data(i),
ra->high_pass_split_data(i),
ra->filter_states(i)->analysis_filter_state1,
ra->filter_states(i)->analysis_filter_state2);
}
}
RETURN_ON_ERR(echo_cancellation_->ProcessRenderAudio(render_audio_.get()));
RETURN_ON_ERR(echo_control_mobile_->ProcessRenderAudio(render_audio_.get()));
RETURN_ON_ERR(gain_control_->ProcessRenderAudio(render_audio_.get()));
RETURN_ON_ERR(echo_cancellation_->ProcessRenderAudio(ra));
RETURN_ON_ERR(echo_control_mobile_->ProcessRenderAudio(ra));
RETURN_ON_ERR(gain_control_->ProcessRenderAudio(ra));
return kNoError;
}