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ACM: Adding support for more than 2 channels in the send pipeline

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This CL adds support in the audio coding module for sending more than
2 channels to the encoder.

Bug: webrtc:11007
Change-Id: I0909b5c37a54c9d2e1353b864e55008cda50ffae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155583
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29385}
This commit is contained in:
Per Åhgren
2019-10-04 11:06:15 +02:00
committed by Commit Bot
parent dc34a25ca4
commit 4f2e9406c9
2 changed files with 179 additions and 44 deletions
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90
modules/audio_coding/acm2/audio_coding_module_unittest.cc
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@ -1634,6 +1634,96 @@ TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
RunInner(40000, 60000);
}
// Verify that it works when the data to send is mono and the encoder is set to
// send surround audio.
TEST_F(AudioCodingModuleTestOldApi, SendingMultiChannelForMonoInput) {
constexpr int kSampleRateHz = 48000;
constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
audio_format_ = SdpAudioFormat({"multiopus",
kSampleRateHz,
6,
{{"minptime", "10"},
{"useinbandfec", "1"},
{"channel_mapping", "0,4,1,2,3,5"},
{"num_streams", "4"},
{"coupled_streams", "2"}}});
RegisterCodec();
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = kSamplesPerChannel;
for (size_t k = 0; k < 10; ++k) {
ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kSamplesPerChannel;
}
}
// Verify that it works when the data to send is stereo and the encoder is set
// to send surround audio.
TEST_F(AudioCodingModuleTestOldApi, SendingMultiChannelForStereoInput) {
constexpr int kSampleRateHz = 48000;
constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
audio_format_ = SdpAudioFormat({"multiopus",
kSampleRateHz,
6,
{{"minptime", "10"},
{"useinbandfec", "1"},
{"channel_mapping", "0,4,1,2,3,5"},
{"num_streams", "4"},
{"coupled_streams", "2"}}});
RegisterCodec();
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 2;
input_frame_.samples_per_channel_ = kSamplesPerChannel;
for (size_t k = 0; k < 10; ++k) {
ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kSamplesPerChannel;
}
}
// Verify that it works when the data to send is mono and the encoder is set to
// send stereo audio.
TEST_F(AudioCodingModuleTestOldApi, SendingStereoForMonoInput) {
constexpr int kSampleRateHz = 48000;
constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
audio_format_ = SdpAudioFormat("opus", kSampleRateHz, 2);
RegisterCodec();
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = kSamplesPerChannel;
for (size_t k = 0; k < 10; ++k) {
ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kSamplesPerChannel;
}
}
// Verify that it works when the data to send is stereo and the encoder is set
// to send mono audio.
TEST_F(AudioCodingModuleTestOldApi, SendingMonoForStereoInput) {
constexpr int kSampleRateHz = 48000;
constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1);
RegisterCodec();
input_frame_.sample_rate_hz_ = kSampleRateHz;
input_frame_.num_channels_ = 1;
input_frame_.samples_per_channel_ = kSamplesPerChannel;
for (size_t k = 0; k < 10; ++k) {
ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
input_frame_.timestamp_ += kSamplesPerChannel;
}
}
// The result on the Android platforms is inconsistent for this test case.
// On android_rel the result is different from android and android arm64 rel.
#if defined(WEBRTC_ANDROID)