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| author | aluebs <aluebs@webrtc.org> | 2016-01-11 20:32:29 -0800 |
|---|---|---|
| committer | Commit bot <commit-bot@chromium.org> | 2016-01-12 04:32:32 +0000 |
| commit | b2328d11dcc86fba1661ee3fa0d51fc126939764 (patch) | |
| tree | 54514d04c1037e1bce85076e3c30ba6c13c469b0 /talk | |
| parent | e93ad1b12913981eaf2c8ba278921a30167bf77f (diff) | |
| download | webrtc-b2328d11dcc86fba1661ee3fa0d51fc126939764.tar.gz | |
Remove additional channel constraints when Beamforming is enabled in AudioProcessing
The general constraints on number of channels for AudioProcessing is:
num_in_channels == num_out_channels || num_out_channels == 1
When Beamforming is enabled and additional constraint was added forcing:
num_out_channels == 1
This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo.
Review URL: https://codereview.webrtc.org/1571013002
Cr-Commit-Position: refs/heads/master@{#11215}
Diffstat (limited to 'talk')
| -rw-r--r-- | talk/media/webrtc/fakewebrtcvoiceengine.h | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h index bf22a290b8..eb3b4d3b18 100644 --- a/talk/media/webrtc/fakewebrtcvoiceengine.h +++ b/talk/media/webrtc/fakewebrtcvoiceengine.h @@ -78,6 +78,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing { WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); WEBRTC_STUB_CONST(num_input_channels, ()); + WEBRTC_STUB_CONST(num_proc_channels, ()); WEBRTC_STUB_CONST(num_output_channels, ()); WEBRTC_STUB_CONST(num_reverse_channels, ()); WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); |