Reland "Simplification and refactoring of the AudioBuffer code"
This is a reland of 81c0cf287c8514cb1cd6f3baca484d668c6eb128
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}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
diff --git a/modules/audio_processing/gain_control_unittest.cc b/modules/audio_processing/gain_control_unittest.cc
index e249a11..8014f8a 100644
--- a/modules/audio_processing/gain_control_unittest.cc
+++ b/modules/audio_processing/gain_control_unittest.cc
@@ -80,16 +80,16 @@
const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
const StreamConfig render_config(sample_rate_hz, num_channels, false);
AudioBuffer render_buffer(
- render_config.num_frames(), render_config.num_channels(),
- render_config.num_frames(), 1, render_config.num_frames());
+ render_config.sample_rate_hz(), render_config.num_channels(),
+ render_config.sample_rate_hz(), 1, render_config.sample_rate_hz(), 1);
test::InputAudioFile render_file(
test::GetApmRenderTestVectorFileName(sample_rate_hz));
std::vector<float> render_input(samples_per_channel * num_channels);
const StreamConfig capture_config(sample_rate_hz, num_channels, false);
AudioBuffer capture_buffer(
- capture_config.num_frames(), capture_config.num_channels(),
- capture_config.num_frames(), 1, capture_config.num_frames());
+ capture_config.sample_rate_hz(), capture_config.num_channels(),
+ capture_config.sample_rate_hz(), 1, capture_config.sample_rate_hz(), 1);
test::InputAudioFile capture_file(
test::GetApmCaptureTestVectorFileName(sample_rate_hz));
std::vector<float> capture_input(samples_per_channel * num_channels);