Revert "Simplification and refactoring of the AudioBuffer code"
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}
diff --git a/modules/audio_processing/gain_controller2_unittest.cc b/modules/audio_processing/gain_controller2_unittest.cc
index 185f2f2..99749cc 100644
--- a/modules/audio_processing/gain_controller2_unittest.cc
+++ b/modules/audio_processing/gain_controller2_unittest.cc
@@ -28,7 +28,8 @@
void SetAudioBufferSamples(float value, AudioBuffer* ab) {
// Sets all the samples in |ab| to |value|.
for (size_t k = 0; k < ab->num_channels(); ++k) {
- std::fill(ab->channels()[k], ab->channels()[k] + ab->num_frames(), value);
+ std::fill(ab->channels_f()[k], ab->channels_f()[k] + ab->num_frames(),
+ value);
}
}
@@ -37,7 +38,7 @@
size_t num_frames,
int sample_rate) {
const int num_samples = rtc::CheckedDivExact(sample_rate, 100);
- AudioBuffer ab(sample_rate, 1, sample_rate, 1, sample_rate);
+ AudioBuffer ab(num_samples, 1, num_samples, 1, num_samples);
// Give time to the level estimator to converge.
for (size_t i = 0; i < num_frames + 1; ++i) {
@@ -46,7 +47,7 @@
}
// Return the last sample from the last processed frame.
- return ab.channels()[0][num_samples - 1];
+ return ab.channels_f()[0][num_samples - 1];
}
AudioProcessing::Config::GainController2 CreateAgc2FixedDigitalModeConfig(
@@ -73,9 +74,9 @@
constexpr size_t kStereo = 2u;
const StreamConfig capture_config(AudioProcessing::kSampleRate48kHz, kStereo,
false);
- AudioBuffer ab(capture_config.sample_rate_hz(), capture_config.num_channels(),
- capture_config.sample_rate_hz(), capture_config.num_channels(),
- capture_config.sample_rate_hz());
+ AudioBuffer ab(capture_config.num_frames(), capture_config.num_channels(),
+ capture_config.num_frames(), capture_config.num_channels(),
+ capture_config.num_frames());
test::InputAudioFile capture_file(
test::GetApmCaptureTestVectorFileName(AudioProcessing::kSampleRate48kHz));
std::vector<float> capture_input(capture_config.num_frames() *
@@ -98,7 +99,7 @@
constexpr float sample_value = 1.f;
SetAudioBufferSamples(sample_value, &ab);
gain_controller->Process(&ab);
- return ab.channels()[0][0];
+ return ab.channels_f()[0][0];
}
} // namespace