commit d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4
author henrik.lundin <henrik.lundin@webrtc.org> Mon Nov 23 14:49:25 2015
committer Commit bot <commit-bot@chromium.org> Mon Nov 23 14:49:31 2015
tree 712b3162831f43763fa384ff7b8572923ed55ce6
parent dfafd124186bcecfd1b3b18c24ce495972e4b50c

NetEq: Add new method last_output_sample_rate_hz

This change moves the logics for keeping track of the last ouput
sample rate from AcmReceiver to NetEq, where it fits better. The
getter function AcmReceiver::current_sample_rate_hz() is renamed to
last_output_sample_rate_hz().

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1467163002

Cr-Commit-Position: refs/heads/master@{#10754}

webrtc/modules/audio_coding/main/acm2/acm_receiver.cc

diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index b4dfe3a..1775029 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -123,7 +123,6 @@
       id_(config.id),
       last_audio_decoder_(nullptr),
       previous_audio_activity_(AudioFrame::kVadPassive),
-      current_sample_rate_hz_(config.neteq_config.sample_rate_hz),
       audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
       last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
       neteq_(NetEq::Create(config.neteq_config)),
@@ -157,9 +156,8 @@
   return neteq_->LeastRequiredDelayMs();
 }
 
-int AcmReceiver::current_sample_rate_hz() const {
-  CriticalSectionScoped lock(crit_sect_.get());
-  return current_sample_rate_hz_;
+int AcmReceiver::last_output_sample_rate_hz() const {
+  return neteq_->last_output_sample_rate_hz();
 }
 
 int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
@@ -224,23 +222,18 @@
     return -1;
   }
 
-  // NetEq always returns 10 ms of audio.
-  current_sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
+  const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
 
   // Update if resampling is required.
-  bool need_resampling = (desired_freq_hz != -1) &&
-      (current_sample_rate_hz_ != desired_freq_hz);
+  const bool need_resampling =
+      (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
 
   if (need_resampling && !resampled_last_output_frame_) {
     // Prime the resampler with the last frame.
     int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
-    int samples_per_channel_int =
-        resampler_.Resample10Msec(last_audio_buffer_.get(),
-                                  current_sample_rate_hz_,
-                                  desired_freq_hz,
-                                  num_channels,
-                                  AudioFrame::kMaxDataSizeSamples,
-                                  temp_output);
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+        num_channels, AudioFrame::kMaxDataSizeSamples, temp_output);
     if (samples_per_channel_int < 0) {
       LOG(LERROR) << "AcmReceiver::GetAudio - "
                      "Resampling last_audio_buffer_ failed.";
@@ -254,13 +247,9 @@
   // TODO(henrik.lundin) Glitches in the output may appear if the output rate
   // from NetEq changes. See WebRTC issue 3923.
   if (need_resampling) {
-    int samples_per_channel_int =
-        resampler_.Resample10Msec(audio_buffer_.get(),
-                                  current_sample_rate_hz_,
-                                  desired_freq_hz,
-                                  num_channels,
-                                  AudioFrame::kMaxDataSizeSamples,
-                                  audio_frame->data_);
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+        num_channels, AudioFrame::kMaxDataSizeSamples, audio_frame->data_);
     if (samples_per_channel_int < 0) {
       LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
       return -1;