AudioCodingModuleImpl::Encode: Use a Buffer instead of a stack-allocated array

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

Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 72de1b1c3dcab0fc0a10a835ec8a6f076271d036..95160d7b5adcce026e2e567afc0046eaabc07c3c 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -157,7 +157,6 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
 AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
 
 int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
-  uint8_t stream[2 * MAX_PAYLOAD_SIZE_BYTE];  // Make room for 1 RED payload.
   AudioEncoder::EncodedInfo encoded_info;
   uint8_t previous_pltype;
 
@@ -179,11 +178,13 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
   last_rtp_timestamp_ = rtp_timestamp;
   first_frame_ = false;
 
-  encoded_info = audio_encoder->Encode(rtp_timestamp, input_data.audio,
-                                       input_data.length_per_channel,
-                                       sizeof(stream), stream);
+  encode_buffer_.SetSize(audio_encoder->MaxEncodedBytes());
+  encoded_info = audio_encoder->Encode(
+      rtp_timestamp, input_data.audio, input_data.length_per_channel,
+      encode_buffer_.size(), encode_buffer_.data());
+  encode_buffer_.SetSize(encoded_info.encoded_bytes);
   bitrate_logger_.MaybeLog(audio_encoder->GetTargetBitrate() / 1000);
-  if (encoded_info.encoded_bytes == 0 && !encoded_info.send_even_if_empty) {
+  if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {

[hlundin-webrtc, 2015/08/25 12:43:43]

Why change this while still using encoded_info.encoded_bytes below? They hold
the same information, but I think it is cleaner if we stick to reading one of
them.

[kwiberg-webrtc, 2015/08/25 13:20:24]

Indeed, that was dismeticulous of me. Fixed (by using encode_buffer_.size()
everywhere).

[hlundin-webrtc, 2015/08/26 13:10:17]

Meticulosity is restored.

     // Not enough data.
     return 0;
   }
@@ -205,7 +206,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
     if (packetization_callback_) {
       packetization_callback_->SendData(
           frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
-          stream, encoded_info.encoded_bytes,
+          encode_buffer_.data(), encode_buffer_.size(),
           my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation
                                                        : nullptr);
     }