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Revert "Simplification and refactoring of the AudioBuffer code"

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This reverts commit 81c0cf287c8514cb1cd6f3baca484d668c6eb128.

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}
This commit is contained in:
Steve Anton
2019-08-21 17:52:28 +00:00
committed by Commit Bot
parent f5815fa6bb
commit f254e9e9e5
32 changed files with 449 additions and 538 deletions
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50
modules/audio_processing/audio_processing_impl.cc
View File

@ -495,17 +495,17 @@ int AudioProcessingImpl::MaybeInitializeRender(
int AudioProcessingImpl::InitializeLocked() {
UpdateActiveSubmoduleStates();
const int render_audiobuffer_sample_rate_hz =
const int render_audiobuffer_num_output_frames =
formats_.api_format.reverse_output_stream().num_frames() == 0
? formats_.render_processing_format.sample_rate_hz()
: formats_.api_format.reverse_output_stream().sample_rate_hz();
? formats_.render_processing_format.num_frames()
: formats_.api_format.reverse_output_stream().num_frames();
if (formats_.api_format.reverse_input_stream().num_channels() > 0) {
render_.render_audio.reset(new AudioBuffer(
formats_.api_format.reverse_input_stream().sample_rate_hz(),
formats_.api_format.reverse_input_stream().num_frames(),
formats_.api_format.reverse_input_stream().num_channels(),
formats_.render_processing_format.sample_rate_hz(),
formats_.render_processing_format.num_frames(),
formats_.render_processing_format.num_channels(),
render_audiobuffer_sample_rate_hz));
render_audiobuffer_num_output_frames));
if (formats_.api_format.reverse_input_stream() !=
formats_.api_format.reverse_output_stream()) {
render_.render_converter = AudioConverter::Create(
@ -521,12 +521,12 @@ int AudioProcessingImpl::InitializeLocked() {
render_.render_converter.reset(nullptr);
}
capture_.capture_audio.reset(new AudioBuffer(
formats_.api_format.input_stream().sample_rate_hz(),
formats_.api_format.input_stream().num_channels(),
capture_nonlocked_.capture_processing_format.sample_rate_hz(),
formats_.api_format.output_stream().num_channels(),
formats_.api_format.output_stream().sample_rate_hz()));
capture_.capture_audio.reset(
new AudioBuffer(formats_.api_format.input_stream().num_frames(),
formats_.api_format.input_stream().num_channels(),
capture_nonlocked_.capture_processing_format.num_frames(),
formats_.api_format.output_stream().num_channels(),
formats_.api_format.output_stream().num_frames()));
AllocateRenderQueue();
@ -1244,11 +1244,11 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
}
capture_.vad_activity = frame->vad_activity_;
capture_.capture_audio->CopyFrom(frame);
capture_.capture_audio->DeinterleaveFrom(frame);
RETURN_ON_ERR(ProcessCaptureStreamLocked());
if (submodule_states_.CaptureMultiBandProcessingActive() ||
submodule_states_.CaptureFullBandProcessingActive()) {
capture_.capture_audio->CopyTo(frame);
capture_.capture_audio->InterleaveTo(frame);
}
frame->vad_activity_ = capture_.vad_activity;
@ -1274,12 +1274,12 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
if (private_submodules_->pre_amplifier) {
private_submodules_->pre_amplifier->ApplyGain(AudioFrameView<float>(
capture_buffer->channels(), capture_buffer->num_channels(),
capture_buffer->channels_f(), capture_buffer->num_channels(),
capture_buffer->num_frames()));
}
capture_input_rms_.Analyze(rtc::ArrayView<const float>(
capture_buffer->channels_const()[0],
capture_buffer->channels_const_f()[0],
capture_nonlocked_.capture_processing_format.num_frames()));
const bool log_rms = ++capture_rms_interval_counter_ >= 1000;
if (log_rms) {
@ -1327,7 +1327,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
if (constants_.use_experimental_agc_process_before_aec) {
private_submodules_->agc_manager->Process(
capture_buffer->channels_const()[0],
capture_buffer->channels_const_f()[0],
capture_nonlocked_.capture_processing_format.num_frames(),
capture_nonlocked_.capture_processing_format.sample_rate_hz());
}
@ -1436,7 +1436,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
if (config_.residual_echo_detector.enabled) {
RTC_DCHECK(private_submodules_->echo_detector);
private_submodules_->echo_detector->AnalyzeCaptureAudio(
rtc::ArrayView<const float>(capture_buffer->channels()[0],
rtc::ArrayView<const float>(capture_buffer->channels_f()[0],
capture_buffer->num_frames()));
}
@ -1449,9 +1449,9 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
: 1.f;
public_submodules_->transient_suppressor->Suppress(
capture_buffer->channels()[0], capture_buffer->num_frames(),
capture_buffer->channels_f()[0], capture_buffer->num_frames(),
capture_buffer->num_channels(),
capture_buffer->split_bands_const(0)[kBand0To8kHz],
capture_buffer->split_bands_const_f(0)[kBand0To8kHz],
capture_buffer->num_frames_per_band(),
capture_.keyboard_info.keyboard_data,
capture_.keyboard_info.num_keyboard_frames, voice_probability,
@ -1474,9 +1474,9 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
}
// The level estimator operates on the recombined data.
public_submodules_->level_estimator->ProcessStream(*capture_buffer);
public_submodules_->level_estimator->ProcessStream(capture_buffer);
if (config_.level_estimation.enabled) {
private_submodules_->output_level_estimator->ProcessStream(*capture_buffer);
private_submodules_->output_level_estimator->ProcessStream(capture_buffer);
capture_.stats.output_rms_dbfs =
private_submodules_->output_level_estimator->RMS();
} else {
@ -1484,7 +1484,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
}
capture_output_rms_.Analyze(rtc::ArrayView<const float>(
capture_buffer->channels_const()[0],
capture_buffer->channels_const_f()[0],
capture_nonlocked_.capture_processing_format.num_frames()));
if (log_rms) {
RmsLevel::Levels levels = capture_output_rms_.AverageAndPeak();
@ -1609,11 +1609,11 @@ int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
aec_dump_->WriteRenderStreamMessage(*frame);
}
render_.render_audio->CopyFrom(frame);
render_.render_audio->DeinterleaveFrom(frame);
RETURN_ON_ERR(ProcessRenderStreamLocked());
if (submodule_states_.RenderMultiBandProcessingActive() ||
submodule_states_.RenderFullBandProcessingActive()) {
render_.render_audio->CopyTo(frame);
render_.render_audio->InterleaveTo(frame);
}
return kNoError;
}