Change opus min bitrate.
BUG=webrtc:7087
Review-Url: https://codereview.webrtc.org/2668693003
Cr-Commit-Position: refs/heads/master@{#16383}
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index bd0afc6..6b39cc8 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -29,7 +29,11 @@
namespace {
constexpr int kSampleRateHz = 48000;
-constexpr int kMinBitrateBps = 500;
+
+// Opus API allows a min bitrate of 500bps, but Opus documentation suggests
+// a minimum bitrate of 6kbps.
+constexpr int kMinBitrateBps = 6000;
+
constexpr int kMaxBitrateBps = 512000;
constexpr int kSupportedFrameLengths[] = {20, 60};
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index ff6e628..11456f2 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -17,9 +17,11 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
+#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
@@ -113,6 +115,23 @@
EXPECT_EQ(*config.num_channels, encoder->num_channels_to_encode());
}
+// Create 10ms audio data blocks for a total packet size of "packet_size_ms".
+std::unique_ptr<test::AudioLoop> Create10msAudioBlocks(
+ const std::unique_ptr<AudioEncoderOpus>& encoder,
+ int packet_size_ms) {
+ const std::string file_name =
+ test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+
+ std::unique_ptr<test::AudioLoop> speech_data(new test::AudioLoop());
+ int audio_samples_per_ms =
+ rtc::CheckedDivExact(encoder->SampleRateHz(), 1000);
+ RTC_DCHECK(speech_data->Init(
+ file_name,
+ packet_size_ms * audio_samples_per_ms * encoder->num_channels_to_encode(),
+ 10 * audio_samples_per_ms * encoder->num_channels_to_encode()));
+ return speech_data;
+}
+
} // namespace
TEST(AudioEncoderOpusTest, DefaultApplicationModeMono) {
@@ -165,7 +184,7 @@
OnReceivedUplinkBandwidthWithoutAudioNetworkAdaptor) {
auto states = CreateCodec(1);
// Constants are replicated from audio_states.encoderopus.cc.
- const int kMinBitrateBps = 500;
+ const int kMinBitrateBps = 6000;
const int kMaxBitrateBps = 512000;
// Set a too low bitrate.
states.encoder->OnReceivedUplinkBandwidth(kMinBitrateBps - 1,
@@ -183,8 +202,8 @@
states.encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps,
rtc::Optional<int64_t>());
EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
- // Set rates from 1000 up to 32000 bps.
- for (int rate = 1000; rate <= 32000; rate += 1000) {
+ // Set rates from kMaxBitrateBps up to 32000 bps.
+ for (int rate = kMinBitrateBps; rate <= 32000; rate += 1000) {
states.encoder->OnReceivedUplinkBandwidth(rate, rtc::Optional<int64_t>());
EXPECT_EQ(rate, states.encoder->GetTargetBitrate());
}
@@ -398,7 +417,7 @@
test::ScopedFieldTrials override_field_trials(
"WebRTC-SendSideBwe-WithOverhead/Enabled/");
- constexpr int kMinBitrateBps = 500;
+ constexpr int kMinBitrateBps = 6000;
constexpr int kMaxBitrateBps = 512000;
auto states = CreateCodec(2);
@@ -499,4 +518,31 @@
}
}
+TEST(AudioEncoderOpusTest, EncodeAtMinBitrate) {
+ auto states = CreateCodec(1);
+ constexpr int kNumPacketsToEncode = 2;
+ auto audio_frames =
+ Create10msAudioBlocks(states.encoder, kNumPacketsToEncode * 20);
+ rtc::Buffer encoded;
+ uint32_t rtp_timestamp = 12345; // Just a number not important to this test.
+
+ states.encoder->OnReceivedUplinkBandwidth(0, rtc::Optional<int64_t>());
+ for (int packet_index = 0; packet_index < kNumPacketsToEncode;
+ packet_index++) {
+ // Make sure we are not encoding before we have enough data for
+ // a 20ms packet.
+ for (int index = 0; index < 1; index++) {
+ states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
+ &encoded);
+ EXPECT_EQ(0u, encoded.size());
+ }
+
+ // Should encode now.
+ states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
+ &encoded);
+ EXPECT_GT(encoded.size(), 0u);
+ encoded.Clear();
+ }
+}
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index deb2a75..1ee92a1 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -608,8 +608,8 @@
namespace {
void TestOpusSetTargetBitrates(AudioEncoder* audio_encoder) {
- EXPECT_EQ(500, SetAndGetTargetBitrate(audio_encoder, 499));
- EXPECT_EQ(500, SetAndGetTargetBitrate(audio_encoder, 500));
+ EXPECT_EQ(6000, SetAndGetTargetBitrate(audio_encoder, 5999));
+ EXPECT_EQ(6000, SetAndGetTargetBitrate(audio_encoder, 6000));
EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder, 32000));
EXPECT_EQ(512000, SetAndGetTargetBitrate(audio_encoder, 512000));
EXPECT_EQ(512000, SetAndGetTargetBitrate(audio_encoder, 513000));
