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Don't always downsample to 16kHz in the reverse stream in APM
The first approach landed here: https://codereview.webrtc.org/1773173002 But it was partially reverted, because it affected the AEC performance, here: https://codereview.webrtc.org/1867483003/ The main difference of this approach is that it doesn't use the 3-band splitting filter in the reverse stream, which seems to be the culprit of the AEC regression. Also, the 2-band splitting filter has been used for the 32kHz case for a long time without any problem, and this is expanded in the CL to cover the 48kHz case as well. BUG=webrtc:5725 TBR=tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1865633005 Cr-Commit-Position: refs/heads/master@{#12451}
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5 changed files with 20 additions and 13 deletions
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@ -366,18 +366,20 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
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std::min(formats_.api_format.input_stream().sample_rate_hz(),
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formats_.api_format.output_stream().sample_rate_hz())));
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// We normally process the reverse stream at 16 kHz. Unless...
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int rev_proc_rate = kSampleRate16kHz;
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int rev_proc_rate = ClosestHigherNativeRate(std::min(
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formats_.api_format.reverse_input_stream().sample_rate_hz(),
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formats_.api_format.reverse_output_stream().sample_rate_hz()));
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// TODO(aluebs): Remove this restriction once we figure out why the 3-band
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// splitting filter degrades the AEC performance.
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if (rev_proc_rate > kSampleRate32kHz) {
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rev_proc_rate = is_rev_processed() ? kSampleRate32kHz : kSampleRate16kHz;
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}
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// If the forward sample rate is 8 kHz, the reverse stream is also processed
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// at this rate.
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if (capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate8kHz) {
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// ...the forward stream is at 8 kHz.
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rev_proc_rate = kSampleRate8kHz;
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} else {
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if (formats_.api_format.reverse_input_stream().sample_rate_hz() ==
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kSampleRate32kHz) {
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// ...or the input is at 32 kHz, in which case we use the splitting
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// filter rather than the resampler.
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rev_proc_rate = kSampleRate32kHz;
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}
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rev_proc_rate = std::max(rev_proc_rate, static_cast<int>(kSampleRate16kHz));
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}
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// Always downmix the reverse stream to mono for analysis. This has been
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@ -1151,11 +1153,11 @@ bool AudioProcessingImpl::is_rev_processed() const {
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bool AudioProcessingImpl::rev_synthesis_needed() const {
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return (is_rev_processed() &&
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formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz);
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is_multi_band(formats_.rev_proc_format.sample_rate_hz()));
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}
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bool AudioProcessingImpl::rev_analysis_needed() const {
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return formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz &&
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return is_multi_band(formats_.rev_proc_format.sample_rate_hz()) &&
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(is_rev_processed() ||
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public_submodules_->echo_cancellation
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->is_enabled_render_side_query() ||
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@ -54,7 +54,12 @@ bool write_ref_data = false;
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const google::protobuf::int32 kChannels[] = {1, 2};
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const int kSampleRates[] = {8000, 16000, 32000, 48000};
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#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
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// Android doesn't support 48kHz.
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const int kProcessSampleRates[] = {8000, 16000, 32000};
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#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
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const int kProcessSampleRates[] = {8000, 16000, 32000, 48000};
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#endif
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enum StreamDirection { kForward = 0, kReverse };
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@ -2692,7 +2697,7 @@ INSTANTIATE_TEST_CASE_P(
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std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0),
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std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20),
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std::tr1::make_tuple(16000, 16000, 48000, 16000, 40, 20),
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std::tr1::make_tuple(16000, 16000, 32000, 16000, 50, 20),
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std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20),
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std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
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#elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
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@ -2748,7 +2753,7 @@ INSTANTIATE_TEST_CASE_P(
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std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0),
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std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20),
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std::tr1::make_tuple(16000, 16000, 48000, 16000, 35, 20),
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std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20),
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std::tr1::make_tuple(16000, 16000, 32000, 16000, 35, 20),
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std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
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#endif
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