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Remove the requirement to call set_sample_rate_hz and friends.

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Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)

Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.

TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.

R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2014-01-07 17:45:09 +00:00
parent 39669c5c8f
commit 60730cfe3c
21 changed files with 308 additions and 466 deletions
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6
webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
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@@ -552,8 +552,8 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
}
if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) {
WebRtcSpl_AnalysisQMF(speech_in_ptr, speechInLB, speechInUB,
instISAC->analysisFBState1,
WebRtcSpl_AnalysisQMF(speech_in_ptr, SWBFRAMESAMPLES_10ms, speechInLB,
speechInUB, instISAC->analysisFBState1,
instISAC->analysisFBState2);
/* Convert from fixed to floating point. */
@@ -1314,7 +1314,7 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
speechIdx = 0;
while (speechIdx < numSamplesLB) {
WebRtcSpl_SynthesisQMF(&outFrameLB[speechIdx], &outFrameUB[speechIdx],
&decoded[(speechIdx << 1)],
FRAMESAMPLES_10ms, &decoded[(speechIdx << 1)],
instISAC->synthesisFBState1,
instISAC->synthesisFBState2);