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Mainly hlundin's patch.

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Review URL: https://webrtc-codereview.appspot.com/1052004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3405 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklas.enbom@webrtc.org
2013-01-24 18:53:43 +00:00
parent 4782911572
commit 05e7bfeeea
12 changed files with 277 additions and 16 deletions
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2
webrtc/modules/audio_coding/main/source/acm_neteq.cc
View File

@ -223,7 +223,7 @@ WebRtc_Word16 ACMNetEQ::AllocatePacketBufferByIdxSafe(
}
if (WebRtcNetEQ_GetRecommendedBufferSize(inst_[idx], used_codecs,
num_codecs,
kTCPLargeJitter,
kTCPXLargeJitter,
&max_num_packets,
&buffer_size_in_bytes) != 0) {
LogError("GetRecommendedBufferSize", idx);

4
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@ -2084,9 +2084,9 @@ int AudioCodingModuleImpl::InitStereoSlave() {
// Minimum playout delay (Used for lip-sync).
WebRtc_Word32 AudioCodingModuleImpl::SetMinimumPlayoutDelay(
const WebRtc_Word32 time_ms) {
if ((time_ms < 0) || (time_ms > 1000)) {
if ((time_ms < 0) || (time_ms > 10000)) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Delay must be in the range of 0-1000 milliseconds.");
"Delay must be in the range of 0-10000 milliseconds.");
return -1;
}
return neteq_.SetExtraDelay(time_ms);