Remove the requirement to call set_sample_rate_hz and friends.

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/webrtc@5346 4adac7df-926f-26a2-2b94-8c16560cd09d

1 files changed, 13 insertions, 46 deletions

diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 2724f528..3f45feab 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -695,10 +695,12 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
     // Store speech type for dead-or-alive detection
     _outputSpeechType = audioFrame.speech_type_;
 
-    // Perform far-end AudioProcessing module processing on the received signal
-    if (_rxApmIsEnabled)
-    {
-        ApmProcessRx(audioFrame);
+    if (_rxApmIsEnabled) {
+      int err = rx_audioproc_->ProcessStream(&audioFrame);
+      if (err) {
+        LOG(LS_ERROR) << "ProcessStream() error: " << err;
+        assert(false);
+      }
     }
 
     float output_gain = 1.0f;
@@ -4446,29 +4448,13 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
 
     InsertInbandDtmfTone();
 
-    if (_includeAudioLevelIndication)
-    {
-        if (rtp_audioproc_->set_sample_rate_hz(_audioFrame.sample_rate_hz_) !=
-            AudioProcessing::kNoError)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                         VoEId(_instanceId, _channelId),
-                         "Error setting AudioProcessing sample rate");
-            return -1;
-        }
-
-        if (rtp_audioproc_->set_num_channels(_audioFrame.num_channels_,
-                                             _audioFrame.num_channels_) !=
-            AudioProcessing::kNoError)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                         VoEId(_instanceId, _channelId),
-                         "Error setting AudioProcessing channels");
-            return -1;
-        }
-
-        // Performs level analysis only; does not affect the signal.
-        rtp_audioproc_->ProcessStream(&_audioFrame);
+    if (_includeAudioLevelIndication) {
+      // Performs level analysis only; does not affect the signal.
+      int err = rtp_audioproc_->ProcessStream(&_audioFrame);
+      if (err) {
+        LOG(LS_ERROR) << "ProcessStream() error: " << err;
+        assert(false);
+      }
     }
 
     return 0;
@@ -5210,25 +5196,6 @@ Channel::RegisterReceiveCodecsToRTPModule()
     }
 }
 
-int Channel::ApmProcessRx(AudioFrame& frame) {
-  // Register the (possibly new) frame parameters.
-  if (rx_audioproc_->set_sample_rate_hz(frame.sample_rate_hz_) != 0) {
-    assert(false);
-    LOG_FERR1(LS_ERROR, set_sample_rate_hz, frame.sample_rate_hz_);
-  }
-  if (rx_audioproc_->set_num_channels(frame.num_channels_,
-                                      frame.num_channels_) != 0) {
-    assert(false);
-    LOG_FERR2(LS_ERROR, set_num_channels, frame.num_channels_,
-              frame.num_channels_);
-  }
-  if (rx_audioproc_->ProcessStream(&frame) != 0) {
-    assert(false);
-    LOG_FERR0(LS_ERROR, ProcessStream);
-  }
-  return 0;
-}
-
 int Channel::SetSecondarySendCodec(const CodecInst& codec,
                                    int red_payload_type) {
   // Sanity check for payload type.