Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/acm2/acm_receiver.cc

Index: webrtc/modules/audio_coding/acm2/acm_receiver.cc
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 39d8a0d175e44aa5432a2997759cf0ae671481f0..02d165a2d5a11a9c72cfc45af7cae96a612696d0 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -18,6 +18,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
@@ -120,14 +121,12 @@ bool IsCng(int codec_id) {
 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
     : last_audio_decoder_(nullptr),
       previous_audio_activity_(AudioFrame::kVadPassive),
-      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);
 }
 
@@ -208,19 +207,11 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
 }
 
 int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
-  enum NetEqOutputType type;
-  size_t samples_per_channel;
-  size_t num_channels;
-
   // Accessing members, take the lock.
   rtc::CritScope lock(&crit_sect_);
 
-  // Always write the output to |audio_buffer_| first.
-  if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
-                       audio_buffer_.get(),
-                       &samples_per_channel,
-                       &num_channels,
-                       &type) != NetEq::kOK) {
+  enum NetEqOutputType type;
+  if (neteq_->GetAudio(audio_frame, &type) != NetEq::kOK) {
     LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
     return -1;
   }
@@ -236,44 +227,42 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_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);
+        audio_frame->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_);
+        audio_frame->data_, current_sample_rate_hz, desired_freq_hz,
+        audio_frame->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);
+    audio_frame->samples_per_channel_ =
+        static_cast<size_t>(samples_per_channel_int);
+    audio_frame->sample_rate_hz_ = desired_freq_hz;
+    RTC_DCHECK_EQ(
+        audio_frame->sample_rate_hz_,
+        rtc::checked_cast<int>(audio_frame->samples_per_channel_ * 100));
     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_,
-           audio_buffer_.get(),
-           samples_per_channel * num_channels * sizeof(int16_t));
   }
 
-  // Swap buffers, so that the current audio is stored in |last_audio_buffer_|
-  // for next time.
-  audio_buffer_.swap(last_audio_buffer_);
-
-  audio_frame->num_channels_ = num_channels;
-  audio_frame->samples_per_channel_ = samples_per_channel;
-  audio_frame->sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
+  // Store current audio in |last_audio_buffer_| for next time.
+  memcpy(last_audio_buffer_.get(), audio_frame->data_,
+         sizeof(int16_t) * audio_frame->samples_per_channel_ *
+             audio_frame->num_channels_);
 
   // Should set |vad_activity| before calling SetAudioFrameActivityAndType().
   audio_frame->vad_activity_ = previous_audio_activity_;
@@ -284,6 +273,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   // Computes the RTP timestamp of the first sample in |audio_frame| from
   // |GetPlayoutTimestamp|, which is the timestamp of the last sample of
   // |audio_frame|.
+  // TODO(henrik.lundin) Move setting of audio_frame->timestamp_ inside NetEq.
   uint32_t playout_timestamp = 0;
   if (GetPlayoutTimestamp(&playout_timestamp)) {
     audio_frame->timestamp_ = playout_timestamp -