diff options
Diffstat (limited to 'modules/audio_coding/main')
| -rw-r--r-- | modules/audio_coding/main/acm2/audio_coding_module_impl.cc | 10 | ||||
| -rw-r--r-- | modules/audio_coding/main/test/opus_test.cc | 18 |
2 files changed, 15 insertions, 13 deletions
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc index f2410b7d..26f5b542 100644 --- a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc +++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc @@ -1618,14 +1618,8 @@ int AudioCodingModuleImpl::ReceiveFrequency() const { int codec_id = receiver_.last_audio_codec_id(); - int sample_rate_hz; - if (codec_id < 0) - sample_rate_hz = receiver_.current_sample_rate_hz(); - else - sample_rate_hz = ACMCodecDB::database_[codec_id].plfreq; - - // TODO(tlegrand): Remove this option when we have full 48 kHz support. - return (sample_rate_hz > 32000) ? 32000 : sample_rate_hz; + return codec_id < 0 ? receiver_.current_sample_rate_hz() : + ACMCodecDB::database_[codec_id].plfreq; } // Get current playout frequency. diff --git a/modules/audio_coding/main/test/opus_test.cc b/modules/audio_coding/main/test/opus_test.cc index 261eb613..398d59da 100644 --- a/modules/audio_coding/main/test/opus_test.cc +++ b/modules/audio_coding/main/test/opus_test.cc @@ -218,6 +218,8 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, int written_samples = 0; int read_samples = 0; int decoded_samples = 0; + bool first_packet = true; + uint32_t start_time_stamp = 0; channel->reset_payload_size(); counter_ = 0; @@ -324,6 +326,10 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, // Send data to the channel. "channel" will handle the loss simulation. channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_, bitstream, bitstream_len_byte, NULL); + if (first_packet) { + first_packet = false; + start_time_stamp = rtp_timestamp_; + } rtp_timestamp_ += frame_length; read_samples += frame_length * channels; } @@ -344,9 +350,11 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, // Write stand-alone speech to file. out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels); - // Number of channels should be the same for both stand-alone and - // ACM-decoding. - EXPECT_EQ(audio_frame.num_channels_, channels); + if (audio_frame.timestamp_ > start_time_stamp) { + // Number of channels should be the same for both stand-alone and + // ACM-decoding. + EXPECT_EQ(audio_frame.num_channels_, channels); + } decoded_samples = 0; } @@ -367,13 +375,13 @@ void OpusTest::OpenOutFile(int test_number) { file_stream << webrtc::test::OutputPath() << "opustest_out_" << test_number << ".pcm"; file_name = file_stream.str(); - out_file_.Open(file_name, 32000, "wb"); + out_file_.Open(file_name, 48000, "wb"); file_stream.str(""); file_name = file_stream.str(); file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" << test_number << ".pcm"; file_name = file_stream.str(); - out_file_standalone_.Open(file_name, 32000, "wb"); + out_file_standalone_.Open(file_name, 48000, "wb"); } } // namespace webrtc |