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Change opus min bitrate.

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BUG=webrtc:7087

Review-Url: https://codereview.webrtc.org/2668693003
Cr-Commit-Position: refs/heads/master@{#16383}
This commit is contained in:
michaelt
2017-01-31 09:06:53 -08:00
committed by Commit bot
parent cf34fdea19
commit 54340d8e75
3 changed files with 57 additions and 7 deletions
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6
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -29,7 +29,11 @@ namespace webrtc {
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};

54
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
View File

@ -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 @@ void CheckEncoderRuntimeConfig(
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 @@ TEST(AudioEncoderOpusTest,
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 @@ TEST(AudioEncoderOpusTest,
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(AudioEncoderOpusTest, BitrateBounded) {
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, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
}
}
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

4
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@ -608,8 +608,8 @@ TEST_F(AudioDecoderOpusTest, EncodeDecode) {
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));