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Add AudioDecoderOpus::MakeAudioDecoder overload taking Environment

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Mark old overload deprecated.
This allows to migrate both calls through AudioDecoderFactory and direct calls to AudioDecpderOpus trait.

Bug: webrtc:356878416
Change-Id: I1502aee5b18aac43a8258e77b770c8e73a056f92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359741
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42793}
This commit is contained in:
Danil Chapovalov 2024-08-16 14:27:43 +02:00 committed by WebRTC LUCI CQ
parent e0fe4200eb
commit 24823c502b
12 changed files with 61 additions and 17 deletions
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7
api/audio_codecs/audio_decoder_factory_template.h
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@ -58,11 +58,16 @@ struct Helper<T, Ts...> {
"absl::optional<T::Config>");
return opt_config ? true : Helper<Ts...>::IsSupportedDecoder(format);
}
// TODO: bugs.webrtc.org/356878416 - Migrate to `Create`
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id) {
auto opt_config = T::SdpToConfig(format);
return opt_config ? T::MakeAudioDecoder(*opt_config, codec_pair_id)
return opt_config ?
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wdeprecated-declarations"
T::MakeAudioDecoder(*opt_config, codec_pair_id)
#pragma clang diagnostic pop
: Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id);
}
};

2
api/audio_codecs/opus/BUILD.gn
View File

@ -67,6 +67,8 @@ rtc_library("audio_decoder_opus") {
"../../../modules/audio_coding:webrtc_opus",
"../../../rtc_base:checks",
"../../../rtc_base/system:rtc_export",
"../../environment",
"../../transport:field_trial_based_config",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]

21
api/audio_codecs/opus/audio_decoder_opus.cc
View File

@ -22,6 +22,7 @@
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/field_trials_view.h"
#include "api/transport/field_trial_based_config.h"
#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include "rtc_base/checks.h"
@ -79,6 +80,17 @@ void AudioDecoderOpus::AppendSupportedDecoders(
specs->push_back({std::move(opus_format), opus_info});
}
std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
const Environment& env,
Config config) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;
}
return std::make_unique<AudioDecoderOpusImpl>(
env.field_trials(), config.num_channels, config.sample_rate_hz);
}
std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
@ -87,8 +99,13 @@ std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
RTC_DCHECK_NOTREACHED();
return nullptr;
}
return std::make_unique<AudioDecoderOpusImpl>(config.num_channels,
config.sample_rate_hz);
if (field_trials == nullptr) {
return std::make_unique<AudioDecoderOpusImpl>(
FieldTrialBasedConfig(), config.num_channels, config.sample_rate_hz);
} else {
return std::make_unique<AudioDecoderOpusImpl>(
*field_trials, config.num_channels, config.sample_rate_hz);
}
}
} // namespace webrtc

9
api/audio_codecs/opus/audio_decoder_opus.h
View File

@ -18,6 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
@ -33,10 +34,16 @@ struct RTC_EXPORT AudioDecoderOpus {
};
static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
[[deprecated("bugs.webrtc.org/356878416")]] //
static std::unique_ptr<AudioDecoder>
MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const FieldTrialsView* field_trials = nullptr);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Environment& env,
Config config);
};
} // namespace webrtc

2
modules/audio_coding/BUILD.gn
View File

@ -485,7 +485,6 @@ rtc_library("webrtc_opus") {
"../../rtc_base:safe_minmax",
"../../rtc_base:stringutils",
"../../rtc_base:timeutils",
"../../system_wrappers:field_trial",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
@ -1207,6 +1206,7 @@ if (rtc_include_tests) {
"../../api/environment:environment_factory",
"../../common_audio",
"../../rtc_base/system:arch",
"../../test:explicit_key_value_config",
"../../test:fileutils",
"../../test:test_main",
"../../test:test_support",

7
modules/audio_coding/codecs/opus/audio_decoder_opus.cc
View File

@ -15,17 +15,18 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/field_trials_view.h"
#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels,
AudioDecoderOpusImpl::AudioDecoderOpusImpl(const FieldTrialsView& field_trials,
size_t num_channels,
int sample_rate_hz)
: channels_(num_channels),
sample_rate_hz_(sample_rate_hz),
generate_plc_(field_trial::IsEnabled("WebRTC-Audio-OpusGeneratePlc")) {
generate_plc_(field_trials.IsEnabled("WebRTC-Audio-OpusGeneratePlc")) {
RTC_DCHECK(num_channels == 1 || num_channels == 2);
RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000);
const int error =

7
modules/audio_coding/codecs/opus/audio_decoder_opus.h
View File

@ -17,6 +17,7 @@
#include <vector>
#include "api/audio_codecs/audio_decoder.h"
#include "api/field_trials_view.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "rtc_base/buffer.h"
@ -24,8 +25,10 @@ namespace webrtc {
class AudioDecoderOpusImpl final : public AudioDecoder {
public:
explicit AudioDecoderOpusImpl(size_t num_channels,
int sample_rate_hz = 48000);
explicit AudioDecoderOpusImpl(const FieldTrialsView& field_trails,
size_t num_channels,
int sample_rate_hz);
~AudioDecoderOpusImpl() override;
AudioDecoderOpusImpl(const AudioDecoderOpusImpl&) = delete;

2
modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
View File

@ -122,7 +122,7 @@ TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) {
// Create decoder.
AudioDecoderOpus::Config dec_config;
dec_config.num_channels = kNumChannels;
auto decoder = AudioDecoderOpus::MakeAudioDecoder(dec_config);
auto decoder = AudioDecoderOpus::MakeAudioDecoder(env, dec_config);
// Open speech file.
const std::string kInputFileName =

7
modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -28,6 +28,7 @@
#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "rtc_base/system/arch.h"
#include "test/explicit_key_value_config.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@ -35,6 +36,8 @@ namespace webrtc {
namespace {
using test::ExplicitKeyValueConfig;
constexpr int kOverheadBytesPerPacket = 50;
// The absolute difference between the input and output (the first channel) is
@ -384,8 +387,8 @@ class AudioDecoderOpusTest
codec_input_rate_hz_ = opus_sample_rate_hz_;
frame_size_ = rtc::CheckedDivExact(opus_sample_rate_hz_, 100);
data_length_ = 10 * frame_size_;
decoder_ =
new AudioDecoderOpusImpl(opus_num_channels_, opus_sample_rate_hz_);
decoder_ = new AudioDecoderOpusImpl(
ExplicitKeyValueConfig(""), opus_num_channels_, opus_sample_rate_hz_);
AudioEncoderOpusConfig config;
config.frame_size_ms = 10;
config.sample_rate_hz = opus_sample_rate_hz_;

2
test/fuzzers/BUILD.gn
View File

@ -330,6 +330,7 @@ webrtc_fuzzer_test("audio_decoder_opus_fuzzer") {
sources = [ "audio_decoder_opus_fuzzer.cc" ]
deps = [
":audio_decoder_fuzzer",
"..:explicit_key_value_config",
"../../modules/audio_coding:webrtc_opus",
]
}
@ -338,6 +339,7 @@ webrtc_fuzzer_test("audio_decoder_opus_redundant_fuzzer") {
sources = [ "audio_decoder_opus_redundant_fuzzer.cc" ]
deps = [
":audio_decoder_fuzzer",
"..:explicit_key_value_config",
"../../modules/audio_coding:webrtc_opus",
]
}

6
test/fuzzers/audio_decoder_opus_fuzzer.cc
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@ -9,13 +9,15 @@
*/
#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include "test/explicit_key_value_config.h"
#include "test/fuzzers/audio_decoder_fuzzer.h"
namespace webrtc {
void FuzzOneInput(const uint8_t* data, size_t size) {
const size_t channels = (size % 2) + 1; // 1 or 2 channels.
AudioDecoderOpusImpl dec(channels);
const int kSampleRateHz = 48000;
const int kSampleRateHz = 48'000;
AudioDecoderOpusImpl dec(test::ExplicitKeyValueConfig(""), channels,
kSampleRateHz);
const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10; // 100 ms.
int16_t output[kAllocatedOuputSizeSamples];
FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec,

6
test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc
View File

@ -9,13 +9,15 @@
*/
#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include "test/explicit_key_value_config.h"
#include "test/fuzzers/audio_decoder_fuzzer.h"
namespace webrtc {
void FuzzOneInput(const uint8_t* data, size_t size) {
const size_t channels = (size % 2) + 1; // 1 or 2 channels.
AudioDecoderOpusImpl dec(channels);
const int kSampleRateHz = 48000;
const int kSampleRateHz = 48'000;
AudioDecoderOpusImpl dec(test::ExplicitKeyValueConfig(""), channels,
kSampleRateHz);
const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10; // 100 ms.
int16_t output[kAllocatedOuputSizeSamples];
FuzzAudioDecoder(DecoderFunctionType::kRedundantDecode, data, size, &dec,