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Remove deprecated Audio Processing APIs

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This change removes the deprecated ChannelLayout versions of ProcessStream
and AnalyzeReverseStream.

Bug: webrtc:5298
Change-Id: I8a7e33e89cffac5eceecd00dfd3c96000643f51b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158529
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29641}
This commit is contained in:
Gustaf Ullberg
2019-10-29 09:30:44 +01:00
committed by Commit Bot
parent 6e4e68852a
commit cb30726646
5 changed files with 61 additions and 147 deletions
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49
modules/audio_processing/audio_processing_impl.cc
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@ -834,38 +834,6 @@ void AudioProcessingImpl::RuntimeSettingEnqueuer::Enqueue(
RTC_LOG(LS_ERROR) << "Cannot enqueue a new runtime setting.";
}
int AudioProcessingImpl::ProcessStream(const float* const* src,
size_t samples_per_channel,
int input_sample_rate_hz,
ChannelLayout input_layout,
int output_sample_rate_hz,
ChannelLayout output_layout,
float* const* dest) {
TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_ChannelLayout");
StreamConfig input_stream;
StreamConfig output_stream;
{
// Access the formats_.api_format.input_stream beneath the capture lock.
// The lock must be released as it is later required in the call
// to ProcessStream(,,,);
rtc::CritScope cs(&crit_capture_);
input_stream = formats_.api_format.input_stream();
output_stream = formats_.api_format.output_stream();
}
input_stream.set_sample_rate_hz(input_sample_rate_hz);
input_stream.set_num_channels(ChannelsFromLayout(input_layout));
input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout));
output_stream.set_sample_rate_hz(output_sample_rate_hz);
output_stream.set_num_channels(ChannelsFromLayout(output_layout));
output_stream.set_has_keyboard(LayoutHasKeyboard(output_layout));
if (samples_per_channel != input_stream.num_frames()) {
return kBadDataLengthError;
}
return ProcessStream(src, input_stream, output_stream, dest);
}
int AudioProcessingImpl::ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
@ -1477,23 +1445,6 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
return kNoError;
}
int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
size_t samples_per_channel,
int sample_rate_hz,
ChannelLayout layout) {
TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_ChannelLayout");
rtc::CritScope cs(&crit_render_);
const StreamConfig reverse_config = {
sample_rate_hz,
ChannelsFromLayout(layout),
LayoutHasKeyboard(layout),
};
if (samples_per_channel != reverse_config.num_frames()) {
return kBadDataLengthError;
}
return AnalyzeReverseStreamLocked(data, reverse_config, reverse_config);
}
int AudioProcessingImpl::AnalyzeReverseStream(
const float* const* data,
const StreamConfig& reverse_config) {