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Opus send rate overflows if over 65 kbps

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The member holding the send rate for Opus had too low resolution for rates above ~65 kbps.

I've added a test that checks if the average rate in a Opus test is in the right range. The test fails before my fix, and now passes.

BUG=3267
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6344 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org 2014-06-05 13:42:51 +00:00
parent b51d3ea593
commit 65d61c3924
2 changed files with 18 additions and 4 deletions
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2
webrtc/modules/audio_coding/main/acm2/acm_opus.h
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@ -47,7 +47,7 @@ class ACMOpus : public ACMGenericCodec {
WebRtcOpusEncInst* encoder_inst_ptr_; WebRtcOpusEncInst* encoder_inst_ptr_;
uint16_t sample_freq_; uint16_t sample_freq_;
uint16_t bitrate_; int32_t bitrate_;
int channels_; int channels_;
bool fec_enabled_; bool fec_enabled_;

20
webrtc/modules/audio_coding/main/test/TestStereo.cc
View File

@ -807,6 +807,8 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
uint32_t time_stamp_diff; uint32_t time_stamp_diff;
channel->reset_payload_size(); channel->reset_payload_size();
int error_count = 0; int error_count = 0;
int variable_bytes = 0;
int variable_packets = 0;
while (1) { while (1) {
// Simulate packet loss by setting |packet_loss_| to "true" in // Simulate packet loss by setting |packet_loss_| to "true" in
@ -838,11 +840,16 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
// Run sender side of ACM // Run sender side of ACM
EXPECT_GT(acm_a_->Process(), -1); EXPECT_GT(acm_a_->Process(), -1);
// Verify that the received packet size matches the settings // Verify that the received packet size matches the settings.
rec_size = channel->payload_size(); rec_size = channel->payload_size();
if ((0 < rec_size) & (rec_size < 65535)) { if ((0 < rec_size) & (rec_size < 65535)) {
// Opus is variable rate, skip this test. if (strcmp(send_codec_name_, "opus") == 0) {
if (strcmp(send_codec_name_, "opus")) { // Opus is a variable rate codec, hence calculate the average packet
// size, and later make sure the average is in the right range.
variable_bytes += rec_size;
variable_packets++;
} else {
// For fixed rate codecs, check that packet size is correct.
if ((rec_size != pack_size_bytes_ * out_channels) if ((rec_size != pack_size_bytes_ * out_channels)
&& (pack_size_bytes_ < 65535)) { && (pack_size_bytes_ < 65535)) {
error_count++; error_count++;
@ -866,6 +873,13 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
EXPECT_EQ(0, error_count); EXPECT_EQ(0, error_count);
// Check that packet size is in the right range for variable rate codecs,
// such as Opus.
if (variable_packets > 0) {
variable_bytes /= variable_packets;
EXPECT_NEAR(variable_bytes, pack_size_bytes_, 3);
}
if (in_file_mono_->EndOfFile()) { if (in_file_mono_->EndOfFile()) {
in_file_mono_->Rewind(); in_file_mono_->Rewind();
} }