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Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes

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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
This commit is contained in:
aluebs@webrtc.org 2014-09-03 13:39:01 +00:00
parent adee8f9242
commit bac072667b
1 changed files with 24 additions and 13 deletions
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37
webrtc/modules/audio_processing/test/unpack.cc
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@ -140,9 +140,12 @@ int do_main(int argc, char* argv[]) {
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 @@ int do_main(int argc, char* argv[]) {
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 @@ int do_main(int argc, char* argv[]) {
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 @@ int do_main(int argc, char* argv[]) {
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 @@ int do_main(int argc, char* argv[]) {
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 @@ int do_main(int argc, char* argv[]) {
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 @@ int do_main(int argc, char* argv[]) {
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 @@ int do_main(int argc, char* argv[]) {
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 @@ int do_main(int argc, char* argv[]) {
}
} 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));
}
}
}