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authorhenrika <henrika@webrtc.org>2017-12-22 13:58:29 +0100
committerCommit Bot <commit-bot@chromium.org>2018-01-16 13:45:54 +0000
commit649a385c94235f72aeec7be3d48b193449e139a2 (patch)
tree828e2432a6ce2b112c413f9e2f398d143540d129 /audio/audio_transport_impl.h
parent7d0427865c844a86a0b0d564175f3b07d5b22747 (diff)
downloadwebrtc-649a385c94235f72aeec7be3d48b193449e139a2.tar.gz
Removes usage of analog AGC.
AGC APIs have recently been removed from the ADM. This CL ensures that usage of the analog part of the AGC is removed as well (since this code is dead today anyhow). Bug: webrtc:7306, webrtc:8598 Change-Id: I144f01cd545e5ff6900707c9308906081914e413 Reviewed-on: https://webrtc-review.googlesource.com/36120 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21640}
Diffstat (limited to 'audio/audio_transport_impl.h')
-rw-r--r--audio/audio_transport_impl.h4
1 files changed, 1 insertions, 3 deletions
diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
index e7de7e9f48..8a316a586c 100644
--- a/audio/audio_transport_impl.h
+++ b/audio/audio_transport_impl.h
@@ -31,8 +31,7 @@ class AudioSendStream;
class AudioTransportImpl : public AudioTransport {
public:
AudioTransportImpl(AudioMixer* mixer,
- AudioProcessing* audio_processing,
- AudioDeviceModule* audio_device_module);
+ AudioProcessing* audio_processing);
~AudioTransportImpl() override;
int32_t RecordedDataIsAvailable(const void* audioSamples,
@@ -82,7 +81,6 @@ class AudioTransportImpl : public AudioTransport {
size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
bool typing_noise_detected_ RTC_GUARDED_BY(capture_lock_) = false;
bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
- AudioDeviceModule* audio_device_module_ = nullptr;
PushResampler<int16_t> capture_resampler_;
voe::AudioLevel audio_level_;
TypingDetection typing_detection_;