webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()
Review-Url: https://codereview.webrtc.org/2320053003 Cr-Commit-Position: refs/heads/master@{#14211}
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@ -10,7 +10,6 @@
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#include "webrtc/modules/audio_processing/audio_processing_impl.h"
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#include <assert.h>
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#include <algorithm>
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#include "webrtc/base/checks.h"
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@ -84,7 +83,7 @@ static bool LayoutHasKeyboard(AudioProcessing::ChannelLayout layout) {
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return true;
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}
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assert(false);
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RTC_NOTREACHED();
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return false;
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}
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@ -693,8 +692,8 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
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MaybeInitializeCapture(processing_config, reinitialization_required));
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}
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rtc::CritScope cs_capture(&crit_capture_);
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assert(processing_config.input_stream().num_frames() ==
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formats_.api_format.input_stream().num_frames());
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RTC_DCHECK_EQ(processing_config.input_stream().num_frames(),
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formats_.api_format.input_stream().num_frames());
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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if (debug_dump_.debug_file->is_open()) {
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@ -1010,8 +1009,8 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
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processing_config.reverse_output_stream() = reverse_output_config;
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RETURN_ON_ERR(MaybeInitializeRender(processing_config));
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assert(reverse_input_config.num_frames() ==
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formats_.api_format.reverse_input_stream().num_frames());
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RTC_DCHECK_EQ(reverse_input_config.num_frames(),
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formats_.api_format.reverse_input_stream().num_frames());
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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if (debug_dump_.debug_file->is_open()) {
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