NetEq: Add new method last_output_sample_rate_hz

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

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 105 matching lines @@
                 *i == RentACodec::CodecId::kCNFB));
 }
 
 }  // namespace
 
 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
     : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       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)),
       vad_enabled_(config.neteq_config.enable_post_decode_vad),
       clock_(config.clock),
       resampled_last_output_frame_(true) {
   assert(clock_);
   memset(audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
   memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
 }
@@ skipping 13 matching lines @@
   if (neteq_->SetMaximumDelay(delay_ms))
     return 0;
   LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
   return -1;
 }
 
 int AcmReceiver::LeastRequiredDelayMs() const {
   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,
                               rtc::ArrayView<const uint8_t> incoming_payload) {
   uint32_t receive_timestamp = 0;
   const RTPHeader* header = &rtp_header.header;  // Just a shorthand.
 
   {
     CriticalSectionScoped lock(crit_sect_.get());
 
@@ skipping 44 matching lines @@
   // Always write the output to |audio_buffer_| first.
   if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
                        audio_buffer_.get(),
                        &samples_per_channel,
                        &num_channels,
                        &type) != NetEq::kOK) {
     LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
     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();

[kwiberg-webrtc, 2015/11/23 13:11:51]

Excellent idea to use const here.

 
   // 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(

[hlundin-webrtc, 2015/11/23 11:54:33]

The only actual change in this block is that the local current_sample_rate_hz is
used instead of current_sample_rate_hz_.

[kwiberg-webrtc, 2015/11/23 13:11:51]

Acknowledged.

+        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.";
       return -1;
     }
     samples_per_channel = static_cast<size_t>(samples_per_channel_int);
   }
 
   // The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
   // through resampling, or through straight memcpy.
   // 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(

[hlundin-webrtc, 2015/11/23 11:54:33]

The only actual change in this block is that the local current_sample_rate_hz is
used instead of current_sample_rate_hz_.

[kwiberg-webrtc, 2015/11/23 13:11:51]

Acknowledged.

+        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;
     }
     samples_per_channel = static_cast<size_t>(samples_per_channel_int);
     resampled_last_output_frame_ = true;
   } else {
     resampled_last_output_frame_ = false;
     // We might end up here ONLY if codec is changed.
     memcpy(audio_frame->data_,
@@ skipping 264 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   CriticalSectionScoped lock(crit_sect_.get());
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc