Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/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.
  */
 
 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 
 #include <stdlib.h>  // malloc
 
 #include <algorithm>  // sort
 #include <vector>
 
 #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"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
@@ skipping 82 matching lines @@
                 *i == RentACodec::CodecId::kCNWB ||
                 *i == RentACodec::CodecId::kCNSWB ||
                 *i == RentACodec::CodecId::kCNFB));
 }
 
 }  // namespace
 
 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);
 }
 
 AcmReceiver::~AcmReceiver() {
   delete neteq_;
 }
 
 int AcmReceiver::SetMinimumDelay(int delay_ms) {
   if (neteq_->SetMinimumDelay(delay_ms))
     return 0;
@@ skipping 60 matching lines @@
       0) {
     LOG(LERROR) << "AcmReceiver::InsertPacket "
                 << static_cast<int>(header->payloadType)
                 << " Failed to insert packet";
     return -1;
   }
   return 0;
 }
 
 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;
   }
 
   const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
 
   // Update if resampling is required.
   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);
+        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_;
   SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
   previous_audio_activity_ = audio_frame->vad_activity_;
   call_stats_.DecodedByNetEq(audio_frame->speech_type_);
 
   // 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 -
         static_cast<uint32_t>(audio_frame->samples_per_channel_);
   } else {
     // Remain 0 until we have a valid |playout_timestamp|.
     audio_frame->timestamp_ = 0;
   }
 
   return 0;
@@ skipping 227 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc