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
diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/interface/module_common_types.h
index 14139ad..9d00de7 100644
--- a/webrtc/modules/interface/module_common_types.h
+++ b/webrtc/modules/interface/module_common_types.h
@@ -667,6 +667,10 @@
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 @@
// 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 @@
};
inline AudioFrame::AudioFrame()
- : id_(-1),
- timestamp_(0),
- ntp_time_ms_(0),
- data_(),
- samples_per_channel_(0),
- sample_rate_hz_(0),
- num_channels_(1),
- speech_type_(kUndefined),
- vad_activity_(kVadUnknown),
- energy_(0xffffffff),
- interleaved_(true) {}
+ : data_() {
+ Reset();
+}
+
+inline void AudioFrame::Reset() {
+ id_ = -1;
+ // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
+ // to an invalid value, or add a new member to indicate invalidity.
+ timestamp_ = 0;
+ ntp_time_ms_ = -1;
+ 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,
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
index b7eb885..04f1f2c 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -43,7 +43,6 @@
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 @@
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 @@
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 @@
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[],
diff --git a/webrtc/voice_engine/utility.h b/webrtc/voice_engine/utility.h
index 127bdba..3820695 100644
--- a/webrtc/voice_engine/utility.h
+++ b/webrtc/voice_engine/utility.h
@@ -25,10 +25,9 @@
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
-// the desired values. Updates |samples_per_channel_| accordingly.
-//
-// On failure, returns -1 and copies |src_frame| to |dst_frame|.
+// Expects |dst_frame| to have its sample rate and channels members set to the
+// desired values. Updates the samples per channel member accordingly. No other
+// members will be changed.
void RemixAndResample(const AudioFrame& src_frame,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
@@ -37,6 +36,11 @@
// 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,
