Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes
The sizes saved in the aecdumps were always the input length, and this is not necessarily true when there is a change in sample rate. But the sample rates dumped are correct, so we can calculate the sizes from them knowing that we use 10ms chunks.
BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7039 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/test/unpack.cc b/webrtc/modules/audio_processing/test/unpack.cc
index a225d58..bb76a2d 100644
--- a/webrtc/modules/audio_processing/test/unpack.cc
+++ b/webrtc/modules/audio_processing/test/unpack.cc
@@ -140,9 +140,12 @@
Event event_msg;
int frame_count = 0;
+ int reverse_samples_per_channel = 0;
+ int input_samples_per_channel = 0;
+ int output_samples_per_channel = 0;
+ int num_reverse_channels = 0;
int num_input_channels = 0;
int num_output_channels = 0;
- int num_reverse_channels = 0;
scoped_ptr<WavFile> reverse_wav_file;
scoped_ptr<WavFile> input_wav_file;
scoped_ptr<WavFile> output_wav_file;
@@ -158,8 +161,12 @@
const ReverseStream msg = event_msg.reverse_stream();
if (msg.has_data()) {
+ // TODO(aluebs): Replace "num_reverse_channels *
+ // reverse_samples_per_channel" with "msg.data().size() /
+ // sizeof(int16_t)" and so on when this fix in audio_processing has made
+ // it into stable: https://webrtc-codereview.appspot.com/15299004/
WriteIntData(reinterpret_cast<const int16_t*>(msg.data().data()),
- msg.data().size() / sizeof(int16_t),
+ num_reverse_channels * reverse_samples_per_channel,
reverse_wav_file.get(),
reverse_pcm_file.get());
} else if (msg.channel_size() > 0) {
@@ -168,7 +175,7 @@
data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
}
WriteFloatData(data.get(),
- msg.channel(0).size() / sizeof(float),
+ reverse_samples_per_channel,
num_reverse_channels,
reverse_wav_file.get(),
reverse_pcm_file.get());
@@ -183,7 +190,7 @@
const Stream msg = event_msg.stream();
if (msg.has_input_data()) {
WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
- msg.input_data().size() / sizeof(int16_t),
+ num_input_channels * input_samples_per_channel,
input_wav_file.get(),
input_pcm_file.get());
} else if (msg.input_channel_size() > 0) {
@@ -192,7 +199,7 @@
data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
}
WriteFloatData(data.get(),
- msg.input_channel(0).size() / sizeof(float),
+ input_samples_per_channel,
num_input_channels,
input_wav_file.get(),
input_pcm_file.get());
@@ -200,7 +207,7 @@
if (msg.has_output_data()) {
WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
- msg.output_data().size() / sizeof(int16_t),
+ num_output_channels * output_samples_per_channel,
output_wav_file.get(),
output_pcm_file.get());
} else if (msg.output_channel_size() > 0) {
@@ -210,7 +217,7 @@
reinterpret_cast<const float*>(msg.output_channel(i).data());
}
WriteFloatData(data.get(),
- msg.output_channel(0).size() / sizeof(float),
+ output_samples_per_channel,
num_output_channels,
output_wav_file.get(),
output_pcm_file.get());
@@ -276,6 +283,10 @@
output_sample_rate = input_sample_rate;
}
+ reverse_samples_per_channel = reverse_sample_rate / 100;
+ input_samples_per_channel = input_sample_rate / 100;
+ output_samples_per_channel = output_sample_rate / 100;
+
if (FLAGS_pcm) {
if (!reverse_pcm_file.get()) {
reverse_pcm_file.reset(new PcmFile(FLAGS_reverse_file));
@@ -288,14 +299,14 @@
}
} else {
reverse_wav_file.reset(new WavFile(FLAGS_reverse_wav_file,
- reverse_sample_rate,
- num_reverse_channels));
+ reverse_sample_rate,
+ num_reverse_channels));
input_wav_file.reset(new WavFile(FLAGS_input_wav_file,
- input_sample_rate,
- num_input_channels));
+ input_sample_rate,
+ num_input_channels));
output_wav_file.reset(new WavFile(FLAGS_output_wav_file,
- output_sample_rate,
- num_output_channels));
+ output_sample_rate,
+ num_output_channels));
}
}
}
