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(2013)

Unified Diff: webrtc/voice_engine/transmit_mixer.cc

Issue 1338833002: Fix the maximum native sample rate in AudioProcessing (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix 44k1 sample rate Created 5 years, 3 months ago
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Index: webrtc/voice_engine/transmit_mixer.cc
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index a02f298509fc521845f8ae602c0886dedaefb620..8b0a479b6fa1b6431b50143c83fc1719fea79cb2 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -1136,10 +1136,6 @@ void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
int codec_rate;
int num_codec_channels;
GetSendCodecInfo(&codec_rate, &num_codec_channels);
- // TODO(ajm): This currently restricts the sample rate to 32 kHz.
- // 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.
int max_sample_rate_hz = kAudioProcMaxNativeSampleRateHz;
if (audioproc_->echo_control_mobile()->is_enabled()) {
// AECM only supports 8 and 16 kHz.

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