AGC: Removing unnneccessary copying and changing to using const
The changes have been shown to be bitexact on a large dataset.
Bug: webrtc:10859
Change-Id: Iedc0e9e944ebfabb717dd7fb4d2682c695da883e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159694
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29883}
diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 81ded91..9f79b54 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -369,10 +369,11 @@
splitting_filter_->Synthesis(split_data_.get(), data_.get());
}
-void AudioBuffer::ExportSplitChannelData(size_t channel,
- int16_t* const* split_band_data) {
+void AudioBuffer::ExportSplitChannelData(
+ size_t channel,
+ int16_t* const* split_band_data) const {
for (size_t k = 0; k < num_bands(); ++k) {
- const float* band_data = split_bands(channel)[k];
+ const float* band_data = split_bands_const(channel)[k];
RTC_DCHECK(split_band_data[k]);
RTC_DCHECK(band_data);
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index d27ccca..161c509 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -124,7 +124,8 @@
void MergeFrequencyBands();
// Copies the split bands data into the integer two-dimensional array.
- void ExportSplitChannelData(size_t channel, int16_t* const* split_band_data);
+ void ExportSplitChannelData(size_t channel,
+ int16_t* const* split_band_data) const;
// Copies the data in the integer two-dimensional array into the split_bands
// data.
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index aaf372e..ff68909 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -1015,7 +1015,7 @@
}
if (!submodules_.agc_manager) {
- GainControlImpl::PackRenderAudioBuffer(audio, &agc_render_queue_buffer_);
+ GainControlImpl::PackRenderAudioBuffer(*audio, &agc_render_queue_buffer_);
// Insert the samples into the queue.
if (!agc_render_signal_queue_->Insert(&agc_render_queue_buffer_)) {
// The data queue is full and needs to be emptied.
@@ -1298,7 +1298,7 @@
submodules_.high_pass_filter->Process(capture_buffer);
}
- RETURN_ON_ERR(submodules_.gain_control->AnalyzeCaptureAudio(capture_buffer));
+ RETURN_ON_ERR(submodules_.gain_control->AnalyzeCaptureAudio(*capture_buffer));
RTC_DCHECK(
!(submodules_.legacy_noise_suppressor && submodules_.noise_suppressor));
if (submodules_.noise_suppressor) {
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 0e17db7..199f378 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -2102,7 +2102,7 @@
std::make_tuple(32000, 16000, 32000, 16000, 25, 20),
std::make_tuple(32000, 16000, 16000, 16000, 25, 0),
- std::make_tuple(16000, 48000, 48000, 48000, 24, 0),
+ std::make_tuple(16000, 48000, 48000, 48000, 23, 0),
std::make_tuple(16000, 48000, 32000, 48000, 24, 30),
std::make_tuple(16000, 48000, 16000, 48000, 24, 20),
std::make_tuple(16000, 44100, 48000, 44100, 15, 20),
@@ -2145,30 +2145,30 @@
std::make_tuple(44100, 16000, 32000, 16000, 19, 20),
std::make_tuple(44100, 16000, 16000, 16000, 19, 0),
- std::make_tuple(32000, 48000, 48000, 48000, 28, 0),
+ std::make_tuple(32000, 48000, 48000, 48000, 27, 0),
std::make_tuple(32000, 48000, 32000, 48000, 65, 30),
std::make_tuple(32000, 48000, 16000, 48000, 30, 20),
std::make_tuple(32000, 44100, 48000, 44100, 20, 20),
std::make_tuple(32000, 44100, 32000, 44100, 20, 15),
std::make_tuple(32000, 44100, 16000, 44100, 20, 15),
- std::make_tuple(32000, 32000, 48000, 32000, 29, 35),
+ std::make_tuple(32000, 32000, 48000, 32000, 27, 35),
std::make_tuple(32000, 32000, 32000, 32000, 0, 0),
- std::make_tuple(32000, 32000, 16000, 32000, 32, 20),
+ std::make_tuple(32000, 32000, 16000, 32000, 30, 20),
std::make_tuple(32000, 16000, 48000, 16000, 20, 20),
std::make_tuple(32000, 16000, 32000, 16000, 20, 20),
std::make_tuple(32000, 16000, 16000, 16000, 20, 0),
- std::make_tuple(16000, 48000, 48000, 48000, 24, 0),
- std::make_tuple(16000, 48000, 32000, 48000, 25, 30),
+ std::make_tuple(16000, 48000, 48000, 48000, 23, 0),
+ std::make_tuple(16000, 48000, 32000, 48000, 24, 30),
std::make_tuple(16000, 48000, 16000, 48000, 25, 20),
std::make_tuple(16000, 44100, 48000, 44100, 15, 20),
std::make_tuple(16000, 44100, 32000, 44100, 15, 15),
std::make_tuple(16000, 44100, 16000, 44100, 15, 15),
std::make_tuple(16000, 32000, 48000, 32000, 24, 35),
- std::make_tuple(16000, 32000, 32000, 32000, 25, 0),
+ std::make_tuple(16000, 32000, 32000, 32000, 24, 0),
std::make_tuple(16000, 32000, 16000, 32000, 25, 20),
- std::make_tuple(16000, 16000, 48000, 16000, 30, 20),
- std::make_tuple(16000, 16000, 32000, 16000, 30, 20),
+ std::make_tuple(16000, 16000, 48000, 16000, 28, 20),
+ std::make_tuple(16000, 16000, 32000, 16000, 28, 20),
std::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
#endif
diff --git a/modules/audio_processing/gain_control_impl.cc b/modules/audio_processing/gain_control_impl.cc
index f0d48b2..7265d7b 100644
--- a/modules/audio_processing/gain_control_impl.cc
+++ b/modules/audio_processing/gain_control_impl.cc
@@ -117,23 +117,22 @@
}
void GainControlImpl::PackRenderAudioBuffer(
- AudioBuffer* audio,
+ const AudioBuffer& audio,
std::vector<int16_t>* packed_buffer) {
- RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
- audio->num_frames_per_band());
+ 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_channels() == 1) {
- FloatS16ToS16(audio->split_bands_const(0)[kBand0To8kHz],
- audio->num_frames_per_band(), mixed_low_pass_data.data());
+ audio.num_frames_per_band());
+ if (audio.num_channels() == 1) {
+ FloatS16ToS16(audio.split_bands_const(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) {
+ 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 =
- FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[0][i]);
+ FloatS16ToS16(audio.split_channels_const(kBand0To8kHz)[0][i]);
for (int j = 1; j < num_channels; ++j) {
- value += FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[j][i]);
+ value += FloatS16ToS16(audio.split_channels_const(kBand0To8kHz)[j][i]);
}
mixed_low_pass_data[i] = value / num_channels;
}
@@ -141,18 +140,17 @@
packed_buffer->clear();
packed_buffer->insert(packed_buffer->end(), mixed_low_pass.data(),
- (mixed_low_pass.data() + audio->num_frames_per_band()));
+ (mixed_low_pass.data() + audio.num_frames_per_band()));
}
-int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
+int GainControlImpl::AnalyzeCaptureAudio(const AudioBuffer& audio) {
if (!enabled_) {
return AudioProcessing::kNoError;
}
RTC_DCHECK(num_proc_channels_);
- RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
- audio->num_frames_per_band());
- RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
+ 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]
@@ -165,13 +163,11 @@
for (auto& gain_controller : gain_controllers_) {
gain_controller->set_capture_level(analog_capture_level_);
- audio->ExportSplitChannelData(capture_channel, split_bands);
+ audio.ExportSplitChannelData(capture_channel, split_bands);
int err =
WebRtcAgc_AddMic(gain_controller->state(), split_bands,
- audio->num_bands(), audio->num_frames_per_band());
-
- audio->ImportSplitChannelData(capture_channel, split_bands);
+ audio.num_bands(), audio.num_frames_per_band());
if (err != AudioProcessing::kNoError) {
return AudioProcessing::kUnspecifiedError;
@@ -183,15 +179,13 @@
for (auto& gain_controller : gain_controllers_) {
int32_t capture_level_out = 0;
- audio->ExportSplitChannelData(capture_channel, split_bands);
+ audio.ExportSplitChannelData(capture_channel, split_bands);
int err =
WebRtcAgc_VirtualMic(gain_controller->state(), split_bands,
- audio->num_bands(), audio->num_frames_per_band(),
+ audio.num_bands(), audio.num_frames_per_band(),
analog_capture_level_, &capture_level_out);
- audio->ImportSplitChannelData(capture_channel, split_bands);
-
gain_controller->set_capture_level(capture_level_out);
if (err != AudioProcessing::kNoError) {
diff --git a/modules/audio_processing/gain_control_impl.h b/modules/audio_processing/gain_control_impl.h
index 7976613..da61c11 100644
--- a/modules/audio_processing/gain_control_impl.h
+++ b/modules/audio_processing/gain_control_impl.h
@@ -36,12 +36,12 @@
~GainControlImpl() override;
void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
- int AnalyzeCaptureAudio(AudioBuffer* audio);
+ int AnalyzeCaptureAudio(const AudioBuffer& audio);
int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
void Initialize(size_t num_proc_channels, int sample_rate_hz);
- static void PackRenderAudioBuffer(AudioBuffer* audio,
+ static void PackRenderAudioBuffer(const AudioBuffer& audio,
std::vector<int16_t>* packed_buffer);
// GainControl implementation.
diff --git a/modules/audio_processing/gain_control_unittest.cc b/modules/audio_processing/gain_control_unittest.cc
index 8014f8a..81e6899 100644
--- a/modules/audio_processing/gain_control_unittest.cc
+++ b/modules/audio_processing/gain_control_unittest.cc
@@ -31,9 +31,9 @@
}
std::vector<int16_t> render_audio;
- GainControlImpl::PackRenderAudioBuffer(render_audio_buffer, &render_audio);
+ GainControlImpl::PackRenderAudioBuffer(*render_audio_buffer, &render_audio);
gain_controller->ProcessRenderAudio(render_audio);
- gain_controller->AnalyzeCaptureAudio(capture_audio_buffer);
+ gain_controller->AnalyzeCaptureAudio(*capture_audio_buffer);
gain_controller->ProcessCaptureAudio(capture_audio_buffer, false);
if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
diff --git a/resources/audio_processing/output_data_fixed.pb.sha1 b/resources/audio_processing/output_data_fixed.pb.sha1
index 421756f..066449e 100644
--- a/resources/audio_processing/output_data_fixed.pb.sha1
+++ b/resources/audio_processing/output_data_fixed.pb.sha1
@@ -1 +1 @@
-a3bf32dd233cbc5009bd4efce12a5e019771cafb
\ No newline at end of file
+f85386d49e89027aa14f2aad36537a8a4e887a61
\ No newline at end of file
diff --git a/resources/audio_processing/output_data_float.pb.sha1 b/resources/audio_processing/output_data_float.pb.sha1
index c654056..341f036 100644
--- a/resources/audio_processing/output_data_float.pb.sha1
+++ b/resources/audio_processing/output_data_float.pb.sha1
@@ -1 +1 @@
-d2c18cded77614146135b1fd1296c4211d22d754
\ No newline at end of file
+734cc6174a5dac2fd87de267fe8d12519fe18321
\ No newline at end of file
diff --git a/test/fuzzers/agc_fuzzer.cc b/test/fuzzers/agc_fuzzer.cc
index 4d97a2d..10de3b8 100644
--- a/test/fuzzers/agc_fuzzer.cc
+++ b/test/fuzzers/agc_fuzzer.cc
@@ -97,12 +97,12 @@
FillAudioBuffer(fuzz_data, &audio);
const bool stream_has_echo = fuzz_data->ReadOrDefaultValue(true);
- gci->AnalyzeCaptureAudio(&audio);
+ gci->AnalyzeCaptureAudio(audio);
gci->ProcessCaptureAudio(&audio, stream_has_echo);
FillAudioBuffer(fuzz_data, &audio);
- gci->PackRenderAudioBuffer(&audio, &packed_render_audio);
+ gci->PackRenderAudioBuffer(audio, &packed_render_audio);
gci->ProcessRenderAudio(packed_render_audio);
}
}