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Revert 3543

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> Changing non-const reference arguments to pointers, ACM
> 
> Part of refactoring of ACM, and recent lint-warnings.
> This CL changes non-const references in the ACM API to pointers.
> 
> BUG=issue1372
> 
> Review URL: https://webrtc-codereview.appspot.com/1103012

TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1116004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3544 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
2013-02-20 15:57:31 +00:00
parent 374aa49e1a
commit eb7ebf20ed
20 changed files with 207 additions and 214 deletions
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10
webrtc/modules/audio_coding/main/test/delay_test.cc
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@ -108,7 +108,7 @@ class DelayTest {
WebRtc_UWord8 num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for(int n = 0; n < num_encoders; n++) {
acm_b_->Codec(n, &my_codec_param);
acm_b_->Codec(n, my_codec_param);
if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
my_codec_param.channels = 1;
else if (my_codec_param.channels > 1)
@ -155,7 +155,7 @@ class DelayTest {
void SendCodec(const CodecConfig& config) {
CodecInst my_codec_param;
ASSERT_EQ(0, AudioCodingModule::Codec(config.name, &my_codec_param,
ASSERT_EQ(0, AudioCodingModule::Codec(config.name, my_codec_param,
config.sample_rate_hz,
config.num_channels));
encoding_sample_rate_hz_ = my_codec_param.plfreq;
@ -201,7 +201,7 @@ class DelayTest {
// Print delay information every 16 frame
if ((num_frames & 0x3F) == 0x3F) {
ACMNetworkStatistics statistics;
acm_b_->NetworkStatistics(&statistics);
acm_b_->NetworkStatistics(statistics);
fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d"
" ts-based average = %6.3f, "
"curr buff-lev = %4u opt buff-lev = %4u \n",
@ -218,11 +218,11 @@ class DelayTest {
in_file_a_.Read10MsData(audio_frame);
ASSERT_EQ(0, acm_a_->Add10MsData(audio_frame));
ASSERT_LE(0, acm_a_->Process());
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, audio_frame));
out_file_b_.Write10MsData(audio_frame.data_,
audio_frame.samples_per_channel_ *
audio_frame.num_channels_);
acm_b_->PlayoutTimestamp(&playout_ts);
acm_b_->PlayoutTimestamp(playout_ts);
received_ts = channel_a2b_->LastInTimestamp();
inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts) /
static_cast<double>(encoding_sample_rate_hz_);