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Fix a data race in ACM1 when audio is pulled.

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BUG=chromium:348511
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6026 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org 2014-04-29 20:27:45 +00:00
parent a86c42c424
commit 9bd49becc1
1 changed files with 10 additions and 11 deletions
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21
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
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@ -2250,18 +2250,10 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
int decoded_seq_num;
uint32_t decoded_timestamp;
bool update_nack =
neteq_.DecodedRtpInfo(&decoded_seq_num, &decoded_timestamp) &&
nack_enabled_; // Update NACK only if it is enabled.
audio_frame->num_channels_ = audio_frame_.num_channels_;
audio_frame->vad_activity_ = audio_frame_.vad_activity_;
audio_frame->speech_type_ = audio_frame_.speech_type_;
neteq_.DecodedRtpInfo(&decoded_seq_num, &decoded_timestamp);
stereo_mode = (audio_frame_.num_channels_ > 1);
// For stereo playout:
// Master and Slave samples are interleaved starting with Master.
const uint16_t receive_freq =
static_cast<uint16_t>(audio_frame_.sample_rate_hz_);
uint16_t receive_freq;
bool tone_detected = false;
int16_t last_detected_tone;
int16_t tone;
@ -2270,10 +2262,17 @@ int32_t AudioCodingModuleImpl::PlayoutData10Ms(
{
CriticalSectionScoped lock(acm_crit_sect_);
audio_frame->num_channels_ = audio_frame_.num_channels_;
audio_frame->vad_activity_ = audio_frame_.vad_activity_;
audio_frame->speech_type_ = audio_frame_.speech_type_;
stereo_mode = (audio_frame_.num_channels_ > 1);
receive_freq = static_cast<uint16_t>(audio_frame_.sample_rate_hz_);
// Update call statistics.
call_stats_.DecodedByNetEq(audio_frame->speech_type_);
if (update_nack) {
if (nack_enabled_ && update_nack) {
assert(nack_.get());
nack_->UpdateLastDecodedPacket(decoded_seq_num, decoded_timestamp);
}