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Remove CodecInst pt.3

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Finally remove CodecInst from common_types.h, including remaining code referencing it.

TBR=kwiberg

Bug: webrtc:7626
Change-Id: I5e6b949ae9093641e33972af8438d1126fc48556
Reviewed-on: https://webrtc-review.googlesource.com/c/114546
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26036}
This commit is contained in:
Fredrik Solenberg
2018-12-17 21:02:22 +01:00
committed by Commit Bot
parent de133ce79e
commit 41f3a43c74
20 changed files with 21 additions and 304 deletions
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17
modules/audio_coding/codecs/opus/audio_encoder_opus.cc
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@ -16,7 +16,6 @@
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "common_types.h"
#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
@ -334,19 +333,6 @@ absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
return absl::nullopt;
}
AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
const CodecInst& codec_inst) {
AudioEncoderOpusConfig config;
config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
config.num_channels = codec_inst.channels;
config.bitrate_bps = codec_inst.rate;
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
return config;
}
absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "opus") ||
@ -494,9 +480,6 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(
SetProjectedPacketLossRate(packet_loss_rate_);
}
AudioEncoderOpusImpl::AudioEncoderOpusImpl(const CodecInst& codec_inst)
: AudioEncoderOpusImpl(CreateConfig(codec_inst), codec_inst.pltype) {}
AudioEncoderOpusImpl::AudioEncoderOpusImpl(int payload_type,
const SdpAudioFormat& format)
: AudioEncoderOpusImpl(*SdpToConfig(format), payload_type) {}

5
modules/audio_coding/codecs/opus/audio_encoder_opus.h
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@ -30,8 +30,6 @@ namespace webrtc {
class RtcEventLog;
struct CodecInst;
class AudioEncoderOpusImpl final : public AudioEncoder {
public:
class NewPacketLossRateOptimizer {
@ -54,8 +52,6 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
RTC_DISALLOW_COPY_AND_ASSIGN(NewPacketLossRateOptimizer);
};
static AudioEncoderOpusConfig CreateConfig(const CodecInst& codec_inst);
// Returns empty if the current bitrate falls within the hysteresis window,
// defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
// Otherwise, returns the current complexity depending on whether the
@ -83,7 +79,6 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
std::unique_ptr<SmoothingFilter> bitrate_smoother);
explicit AudioEncoderOpusImpl(const CodecInst& codec_inst);
AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format);
~AudioEncoderOpusImpl() override;

40
modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
View File

@ -15,7 +15,6 @@
#include "absl/memory/memory.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "common_audio/mocks/mock_smoothing_filter.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
@ -33,21 +32,11 @@ using ::testing::Return;
namespace {
const CodecInst kDefaultOpusSettings = {105, "opus", 48000, 960, 1, 32000};
constexpr int kDefaultOpusPayloadType = 105;
constexpr int kDefaultOpusRate = 32000;
constexpr int kDefaultOpusPacSize = 960;
constexpr int64_t kInitialTimeUs = 12345678;
AudioEncoderOpusConfig CreateConfig(const CodecInst& codec_inst) {
AudioEncoderOpusConfig config;
config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
config.num_channels = codec_inst.channels;
config.bitrate_bps = codec_inst.rate;
config.application = config.num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
return config;
}
AudioEncoderOpusConfig CreateConfigWithParameters(
const SdpAudioFormat::Parameters& params) {
const SdpAudioFormat format("opus", 48000, 2, params);
@ -79,15 +68,22 @@ std::unique_ptr<AudioEncoderOpusStates> CreateCodec(size_t num_channels) {
return adaptor;
};
CodecInst codec_inst = kDefaultOpusSettings;
codec_inst.channels = num_channels;
states->config = CreateConfig(codec_inst);
AudioEncoderOpusConfig config;
config.frame_size_ms = rtc::CheckedDivExact(kDefaultOpusPacSize, 48);
config.num_channels = num_channels;
config.bitrate_bps = kDefaultOpusRate;
config.application = num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
states->config = config;
std::unique_ptr<MockSmoothingFilter> bitrate_smoother(
new MockSmoothingFilter());
states->mock_bitrate_smoother = bitrate_smoother.get();
states->encoder.reset(new AudioEncoderOpusImpl(
states->config, codec_inst.pltype, std::move(creator),
states->config, kDefaultOpusPayloadType, std::move(creator),
std::move(bitrate_smoother)));
return states;
}
@ -436,12 +432,12 @@ TEST(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
auto states = CreateCodec(2);
states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusSettings.rate * 2,
states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusRate * 2,
absl::nullopt);
// Since |OnReceivedOverhead| has not been called, the codec bitrate should
// not change.
EXPECT_EQ(kDefaultOpusSettings.rate, states->encoder->GetTargetBitrate());
EXPECT_EQ(kDefaultOpusRate, states->encoder->GetTargetBitrate());
}
TEST(AudioEncoderOpusTest, OverheadRemovedFromTargetAudioBitrate) {
@ -456,7 +452,7 @@ TEST(AudioEncoderOpusTest, OverheadRemovedFromTargetAudioBitrate) {
constexpr int kTargetBitrateBps = 40000;
states->encoder->OnReceivedUplinkBandwidth(kTargetBitrateBps, absl::nullopt);
int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusPacSize);
EXPECT_EQ(kTargetBitrateBps -
8 * static_cast<int>(kOverheadBytesPerPacket) * packet_rate,
states->encoder->GetTargetBitrate());
@ -474,7 +470,7 @@ TEST(AudioEncoderOpusTest, BitrateBounded) {
constexpr size_t kOverheadBytesPerPacket = 64;
states->encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusPacSize);
// Set a target rate that is smaller than |kMinBitrateBps| when overhead is
// subtracted. The eventual codec rate should be bounded by |kMinBitrateBps|.