Add a Reset() method to AudioFrame.

This method is introduced to try to avoid inconsistent resetting of
AudioFrame members to default/uninitialized values.

Use it at the call points of DownConvertToCodecFormat(). Results in the
following minor functional changes:
- speech_activity_ is set to its uninitialized value. AFAICT, this
member isn't used at all in the capture path.
- timestamp_ is switched from -1 to 0. This member doesn't appear to be
used either in the capture path, but left a TODO for wu to change the
default value to better represent the uninitialized state.

Bonus: Don't copy the frame on error in RemixAndResample(). An error
indicates a logical fault (as pointed out by the asserts) that we should
not attempt to recover from.
BUG=3111
R=turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21519007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6289 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/interface/module_common_types.h

@@ -667,6 +667,10 @@ class AudioFrame {
   AudioFrame();
   virtual ~AudioFrame() {}
 
+  // Resets all members to their default state (except does not modify the
+  // contents of |data_|).
+  void Reset();
+
   // |interleaved_| is not changed by this method.
   void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
                    int samples_per_channel, int sample_rate_hz,
@@ -687,6 +691,7 @@ class AudioFrame {
   // RTP timestamp of the first sample in the AudioFrame.
   uint32_t timestamp_;
   // NTP time of the estimated capture time in local timebase in milliseconds.
+  // -1 represents an uninitialized value.
   int64_t ntp_time_ms_;
   int16_t data_[kMaxDataSizeSamples];
   int samples_per_channel_;
@@ -706,17 +711,24 @@ class AudioFrame {
 };
 
 inline AudioFrame::AudioFrame()
-    : id_(-1),
+    : data_() {
-      timestamp_(0),
+  Reset();
-      ntp_time_ms_(0),
+}
-      data_(),
+
-      samples_per_channel_(0),
+inline void AudioFrame::Reset() {
-      sample_rate_hz_(0),
+  id_ = -1;
-      num_channels_(1),
+  // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
-      speech_type_(kUndefined),
+  // to an invalid value, or add a new member to indicate invalidity.
-      vad_activity_(kVadUnknown),
+  timestamp_ = 0;
-      energy_(0xffffffff),
+  ntp_time_ms_ = -1;
-      interleaved_(true) {}
+  samples_per_channel_ = 0;
+  sample_rate_hz_ = 0;
+  num_channels_ = 0;
+  speech_type_ = kUndefined;
+  vad_activity_ = kVadUnknown;
+  energy_ = 0xffffffff;
+  interleaved_ = true;
+}
 
 inline void AudioFrame::UpdateFrame(int id, uint32_t timestamp,
                                     const int16_t* data,

webrtc/voice_engine/utility.cc

@@ -43,7 +43,6 @@ void RemixAndResample(const AudioFrame& src_frame,
   if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_,
                                     dst_frame->sample_rate_hz_,
                                     audio_ptr_num_channels) == -1) {
-    dst_frame->CopyFrom(src_frame);
     LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_,
               dst_frame->sample_rate_hz_, audio_ptr_num_channels);
     assert(false);
@@ -54,7 +53,6 @@ void RemixAndResample(const AudioFrame& src_frame,
   int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
                                        AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
-    dst_frame->CopyFrom(src_frame);
     LOG_FERR3(LS_ERROR, Resample, audio_ptr, src_length, dst_frame->data_);
     assert(false);
   }
@@ -81,6 +79,7 @@ void DownConvertToCodecFormat(const int16_t* src_data,
   assert(samples_per_channel <= kMaxMonoDataSizeSamples);
   assert(num_channels == 1 || num_channels == 2);
   assert(codec_num_channels == 1 || codec_num_channels == 2);
+  dst_af->Reset();
 
   // Never upsample the capture signal here. This should be done at the
   // end of the send chain.
@@ -116,9 +115,6 @@ void DownConvertToCodecFormat(const int16_t* src_data,
   dst_af->samples_per_channel_ = out_length / num_channels;
   dst_af->sample_rate_hz_ = destination_rate;
   dst_af->num_channels_ = num_channels;
-  dst_af->timestamp_ = -1;
-  dst_af->speech_type_ = AudioFrame::kNormalSpeech;
-  dst_af->vad_activity_ = AudioFrame::kVadUnknown;
 }
 
 void MixWithSat(int16_t target[],

webrtc/voice_engine/utility.h

@@ -25,10 +25,9 @@ class AudioFrame;
 namespace voe {
 
 // Upmix or downmix and resample the audio in |src_frame| to |dst_frame|.
-// Expects |dst_frame| to have its |num_channels_| and |sample_rate_hz_| set to
+// Expects |dst_frame| to have its sample rate and channels members set to the
-// the desired values. Updates |samples_per_channel_| accordingly.
+// desired values. Updates the samples per channel member accordingly. No other
-//
+// members will be changed.
-// On failure, returns -1 and copies |src_frame| to |dst_frame|.
 void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame);
@@ -37,6 +36,11 @@ void RemixAndResample(const AudioFrame& src_frame,
 // specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
 // temporary space and must be of sufficient size to hold the downmixed source
 // audio (recommend using a size of kMaxMonoDataSizeSamples).
+//
+// |dst_af| will have its data and format members (sample rate, channels and
+// samples per channel) set appropriately. No other members will be changed.
+// TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as
+// it shouldn't be needed.
 void DownConvertToCodecFormat(const int16_t* src_data,
                               int samples_per_channel,
                               int num_channels,