acm_corrected_temporary_buffer_size.patch

From 0000000000000000000000000000000000000000 Mon Sep 17 00:00:00 2001
From: Pedro Pontes <pepontes@microsoft.com>
Date: Wed, 8 Apr 2020 23:15:08 +0200
Subject: ACM: Corrected temporary buffer size

    This CL corrects the temporary buffers size in the
    pre-processing of the capture audio before encoding.

    As part of this it removes the ACM-specific hardcoding
    of the size and instead ensures that the size of the
    temporary buffer matches that of the AudioFrame.

    Backports: https://webrtc.googlesource.com/src/+/d82a02c837d33cdfd75121e40dcccd32515e42d6

diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 4911dfdd5360fcbae1c7a783c65d2258ff0b8e61..aecb20ac43ea2c1a7d44c2b1b037e1e4f22498c6 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -547,17 +547,22 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
 
   *ptr_out = &preprocess_frame_;
   preprocess_frame_.num_channels_ = in_frame.num_channels_;
-  int16_t audio[WEBRTC_10MS_PCM_AUDIO];
-  const int16_t* src_ptr_audio = in_frame.data();
+  std::array<int16_t, AudioFrame::kMaxDataSizeSamples> audio;
+  const int16_t* src_ptr_audio;  
   if (down_mix) {
-    // If a resampling is required the output of a down-mix is written into a
+    // If a resampling is required, the output of a down-mix is written into a
     // local buffer, otherwise, it will be written to the output frame.
     int16_t* dest_ptr_audio =
-        resample ? audio : preprocess_frame_.mutable_data();
-    DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio);
+        resample ? audio.data() : preprocess_frame_.mutable_data();
+    RTC_DCHECK_GE(audio.size(), preprocess_frame_.samples_per_channel_);
+    DownMix(in_frame, AudioFrame::kMaxDataSizeSamples, dest_ptr_audio);
     preprocess_frame_.num_channels_ = 1;
-    // Set the input of the resampler is the down-mixed signal.
-    src_ptr_audio = audio;
+    
+    // Set the input of the resampler to the down-mixed signal.
+    src_ptr_audio = audio.data();
+  } else {
+    // Set the input of the resampler to the original data.
+    src_ptr_audio = in_frame.data();
   }
 
   preprocess_frame_.timestamp_ = expected_codec_ts_;
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index da8ffb5a799f1a444aa263d31ee87a9f00bdd35b..7c0226e4f2b2dfad12d7726467a678e1695d6cf3 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -32,8 +32,6 @@ class AudioEncoder;
 class AudioFrame;
 struct RTPHeader;
 
-#define WEBRTC_10MS_PCM_AUDIO 960  // 16 bits super wideband 48 kHz
-
 // Callback class used for sending data ready to be packetized
 class AudioPacketizationCallback {
  public: