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audioproc_f: input AEC dump as string, output audio to vector

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This CL adds the following options:

pass an input AEC dump as a string (currently, the tool can only accept a path to an AEC dump file)
write the processed capture samples to a given vector

Bug: webrtc:10808
Change-Id: I02863c97ec3cd8c03ade2ea8521836f2e7417050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145208
Commit-Queue: Sonia-Florina Horchidan <soniahorchidan@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28826}
This commit is contained in:
Sonia-Florina Horchidan
2019-08-12 09:57:01 +02:00
committed by Commit Bot
parent 81df62b456
commit b75d14c802
13 changed files with 201 additions and 51 deletions
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18
modules/audio_processing/test/test_utils.cc
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@ -68,6 +68,24 @@ void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
file_->WriteSamples(&interleaved_[0], interleaved_.size());
}
ChannelBufferVectorWriter::ChannelBufferVectorWriter(std::vector<float>* output)
: output_(output) {
RTC_DCHECK(output_);
}
ChannelBufferVectorWriter::~ChannelBufferVectorWriter() = default;
void ChannelBufferVectorWriter::Write(const ChannelBuffer<float>& buffer) {
// Account for sample rate changes throughout a simulation.
interleaved_buffer_.resize(buffer.size());
Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
interleaved_buffer_.data());
size_t old_size = output_->size();
output_->resize(old_size + interleaved_buffer_.size());
FloatToFloatS16(interleaved_buffer_.data(), interleaved_buffer_.size(),
output_->data() + old_size);
}
void WriteIntData(const int16_t* data,
size_t length,
WavWriter* wav_file,