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| author | andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-04-30 18:58:23 +0000 |
|---|---|---|
| committer | andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2014-04-30 18:58:23 +0000 |
| commit | c1878acc3e95341cb3c183ffadd2cec9ebe6e149 (patch) | |
| tree | 26472cb66c1d31b1f1c75e2243f94985812062f2 /voice_engine/transmit_mixer.cc | |
| parent | 785c2fdd22c9ad2d172c0976d1224bbe44073bc7 (diff) | |
| download | webrtc-c1878acc3e95341cb3c183ffadd2cec9ebe6e149.tar.gz | |
Only clamp to 16 kHz when AECM is enabled.
Otherwise we could needlessly downsample to 16 kHz (rather than 32 kHz)
when HW AEC is used.
BUG=3259
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6033 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'voice_engine/transmit_mixer.cc')
| -rw-r--r-- | voice_engine/transmit_mixer.cc | 7 |
1 files changed, 6 insertions, 1 deletions
diff --git a/voice_engine/transmit_mixer.cc b/voice_engine/transmit_mixer.cc index cce2f946..54faec91 100644 --- a/voice_engine/transmit_mixer.cc +++ b/voice_engine/transmit_mixer.cc @@ -1173,7 +1173,12 @@ void TransmitMixer::GenerateAudioFrame(const int16_t* audio, // See: https://code.google.com/p/webrtc/issues/detail?id=3146 // When 48 kHz is supported natively by AudioProcessing, this will have // to be changed to handle 44.1 kHz. - codec_rate = std::min(codec_rate, kAudioProcMaxNativeSampleRateHz); + int max_sample_rate_hz = kAudioProcMaxNativeSampleRateHz; + if (audioproc_->echo_control_mobile()->is_enabled()) { + // AECM only supports 8 and 16 kHz. + max_sample_rate_hz = 16000; + } + codec_rate = std::min(codec_rate, max_sample_rate_hz); stereo_codec_ = num_codec_channels == 2; if (!mono_buffer_.get()) { |