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Removing all external access to the integer sample data in AudioBuffer

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This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.

The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.

Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
This commit is contained in:
Per Åhgren
2019-08-20 09:19:21 +02:00
committed by Commit Bot
parent 93d4c10ffc
commit 928146f546
20 changed files with 298 additions and 124 deletions
Download Patch File Download Diff File Expand all files Collapse all files

70
modules/audio_processing/gain_control_impl.cc
View File

@ -118,25 +118,25 @@ void GainControlImpl::ProcessRenderAudio(
void GainControlImpl::PackRenderAudioBuffer(
AudioBuffer* audio,
std::vector<int16_t>* packed_buffer) {
RTC_DCHECK_GE(160, audio->num_frames_per_band());
std::array<int16_t, 160> mixed_low_pass_data;
rtc::ArrayView<const int16_t> mixed_low_pass;
RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
audio->num_frames_per_band());
std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> mixed_low_pass_data;
rtc::ArrayView<const int16_t> mixed_low_pass(mixed_low_pass_data.data(),
audio->num_frames_per_band());
if (audio->num_proc_channels() == 1) {
mixed_low_pass =
rtc::ArrayView<const int16_t>(audio->split_bands_const(0)[kBand0To8kHz],
audio->num_frames_per_band());
FloatS16ToS16(audio->split_bands_const_f(0)[kBand0To8kHz],
audio->num_frames_per_band(), mixed_low_pass_data.data());
} else {
const int num_channels = static_cast<int>(audio->num_channels());
for (size_t i = 0; i < audio->num_frames_per_band(); ++i) {
int32_t value = audio->split_channels_const(kBand0To8kHz)[0][i];
int32_t value =
FloatS16ToS16(audio->split_channels_const_f(kBand0To8kHz)[0][i]);
for (int j = 1; j < num_channels; ++j) {
value += audio->split_channels_const(kBand0To8kHz)[j][i];
value +=
FloatS16ToS16(audio->split_channels_const_f(kBand0To8kHz)[j][i]);
}
mixed_low_pass_data[i] = value / num_channels;
}
mixed_low_pass = rtc::ArrayView<const int16_t>(
mixed_low_pass_data.data(), audio->num_frames_per_band());
}
packed_buffer->clear();
@ -150,17 +150,28 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
}
RTC_DCHECK(num_proc_channels_);
RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
audio->num_frames_per_band());
RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
RTC_DCHECK_LE(*num_proc_channels_, gain_controllers_.size());
int16_t split_band_data[AudioBuffer::kMaxNumBands]
[AudioBuffer::kMaxSplitFrameLength];
int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
split_band_data[0], split_band_data[1], split_band_data[2]};
if (mode_ == kAdaptiveAnalog) {
int capture_channel = 0;
for (auto& gain_controller : gain_controllers_) {
gain_controller->set_capture_level(analog_capture_level_);
int err = WebRtcAgc_AddMic(
gain_controller->state(), audio->split_bands(capture_channel),
audio->num_bands(), audio->num_frames_per_band());
audio->CopySplitChannelDataTo(capture_channel, split_bands);
int err =
WebRtcAgc_AddMic(gain_controller->state(), split_bands,
audio->num_bands(), audio->num_frames_per_band());
audio->CopySplitChannelDataFrom(capture_channel, split_bands);
if (err != AudioProcessing::kNoError) {
return AudioProcessing::kUnspecifiedError;
@ -171,10 +182,15 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
int capture_channel = 0;
for (auto& gain_controller : gain_controllers_) {
int32_t capture_level_out = 0;
int err = WebRtcAgc_VirtualMic(
gain_controller->state(), audio->split_bands(capture_channel),
audio->num_bands(), audio->num_frames_per_band(),
analog_capture_level_, &capture_level_out);
audio->CopySplitChannelDataTo(capture_channel, split_bands);
int err =
WebRtcAgc_VirtualMic(gain_controller->state(), split_bands,
audio->num_bands(), audio->num_frames_per_band(),
analog_capture_level_, &capture_level_out);
audio->CopySplitChannelDataFrom(capture_channel, split_bands);
gain_controller->set_capture_level(capture_level_out);
@ -199,7 +215,8 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
}
RTC_DCHECK(num_proc_channels_);
RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
audio->num_frames_per_band());
RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
stream_is_saturated_ = false;
@ -208,15 +225,22 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
int32_t capture_level_out = 0;
uint8_t saturation_warning = 0;
int16_t split_band_data[AudioBuffer::kMaxNumBands]
[AudioBuffer::kMaxSplitFrameLength];
int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
split_band_data[0], split_band_data[1], split_band_data[2]};
audio->CopySplitChannelDataTo(capture_channel, split_bands);
// The call to stream_has_echo() is ok from a deadlock perspective
// as the capture lock is allready held.
int err = WebRtcAgc_Process(
gain_controller->state(), audio->split_bands_const(capture_channel),
audio->num_bands(), audio->num_frames_per_band(),
audio->split_bands(capture_channel),
gain_controller->state(), split_bands, audio->num_bands(),
audio->num_frames_per_band(), split_bands,
gain_controller->get_capture_level(), &capture_level_out,
stream_has_echo, &saturation_warning);
audio->CopySplitChannelDataFrom(capture_channel, split_bands);
if (err != AudioProcessing::kNoError) {
return AudioProcessing::kUnspecifiedError;
}