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Add floating point support for writing and reading wav files

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This CL adds support for reading and writing floating point
wav files in WebRTC. It also updates the former wav handling
code as well as adds some simplifications.

Beyond this, the CL also adds support in the APM data_dumper
and in the audioproc_f tool for using the floating point wav
format.

Bug: webrtc:11307
Change-Id: I2ea33fd12f590b6031ac85f75708f6cc88a266b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162902
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30423}
This commit is contained in:
Per Åhgren
2020-01-28 09:08:11 +01:00
committed by Commit Bot
parent 190539717b
commit 5dca3f1336
11 changed files with 685 additions and 360 deletions
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1
modules/audio_processing/test/audio_processing_simulator.h
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@ -93,6 +93,7 @@ struct SimulationSettings {
bool store_intermediate_output = false;
bool print_aec_parameter_values = false;
bool dump_internal_data = false;
WavFile::SampleFormat wav_output_format = WavFile::SampleFormat::kInt16;
absl::optional<std::string> dump_internal_data_output_dir;
absl::optional<std::string> call_order_input_filename;
absl::optional<std::string> call_order_output_filename;