Opus implementation of the AudioEncoderFactoryTemplate API
This was previously reverted, because external projects were using the
internal webrtc::AudioEncoderOpus class and broke when it was renamed.
This re-land avoids renaming it immediately, to give those projects
time to adapt. It also has to revert some of the changes I had made to the
Config struct, since that was also used by the same external projects.
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2948483002
Cr-Commit-Position: refs/heads/master@{#18852}
diff --git a/webrtc/api/DEPS b/webrtc/api/DEPS
index e48b568..b0493fa 100644
--- a/webrtc/api/DEPS
+++ b/webrtc/api/DEPS
@@ -21,6 +21,13 @@
"+webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
],
+ # Needed because AudioEncoderOpus is in the wrong place for
+ # backwards compatibilty reasons. See
+ # https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+ "audio_encoder_opus\.h": [
+ "+webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h",
+ ],
+
# We allow .cc files in webrtc/api/ to #include a bunch of stuff
# that's off-limits for the .h files. That's because .h files leak
# their #includes to whoever's #including them, but .cc files do not
diff --git a/webrtc/api/audio_codecs/opus/BUILD.gn b/webrtc/api/audio_codecs/opus/BUILD.gn
new file mode 100644
index 0000000..b82f496
--- /dev/null
+++ b/webrtc/api/audio_codecs/opus/BUILD.gn
@@ -0,0 +1,42 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+if (is_android) {
+ import("//build/config/android/config.gni")
+ import("//build/config/android/rules.gni")
+}
+
+rtc_static_library("audio_encoder_opus_config") {
+ sources = [
+ "audio_encoder_opus_config.cc",
+ "audio_encoder_opus_config.h",
+ ]
+ deps = [
+ "../../../base:rtc_base_approved",
+ ]
+ defines = []
+ if (rtc_opus_variable_complexity) {
+ defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
+ } else {
+ defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
+ }
+}
+
+rtc_source_set("audio_encoder_opus") {
+ sources = [
+ "audio_encoder_opus.h",
+ ]
+ deps = [
+ ":audio_encoder_opus_config",
+ "..:audio_codecs_api",
+ "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed?
+ "../../../base:rtc_base_approved",
+ "../../../modules/audio_coding:webrtc_opus",
+ ]
+}
diff --git a/webrtc/api/audio_codecs/opus/audio_encoder_opus.h b/webrtc/api/audio_codecs/opus/audio_encoder_opus.h
new file mode 100644
index 0000000..4763f44
--- /dev/null
+++ b/webrtc/api/audio_codecs/opus/audio_encoder_opus.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+
+namespace webrtc {
+
+// Opus encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+//
+// NOTE: At the moment, this struct actually resides in another file. This is a
+// temporary backwards compatibility hack; see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+//
+// NOTE: This struct is still under development and may change without notice.
+/*
+struct AudioEncoderOpus {
+ static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
+ const SdpAudioFormat& audio_format);
+ static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+ static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+ static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+ const AudioEncoderOpusConfig&,
+ int payload_type);
+};
+*/
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
diff --git a/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
new file mode 100644
index 0000000..7d29883
--- /dev/null
+++ b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
@@ -0,0 +1,70 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h"
+
+namespace webrtc {
+
+namespace {
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
+// If we are on Android, iOS and/or ARM, use a lower complexity setting by
+// default, to save encoder complexity.
+constexpr int kDefaultComplexity = 5;
+#else
+constexpr int kDefaultComplexity = 9;
+#endif
+
+constexpr int kDefaultLowRateComplexity =
+ WEBRTC_OPUS_VARIABLE_COMPLEXITY ? 9 : kDefaultComplexity;
+
+} // namespace
+
+constexpr int AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+constexpr int AudioEncoderOpusConfig::kMinBitrateBps;
+constexpr int AudioEncoderOpusConfig::kMaxBitrateBps;
+
+AudioEncoderOpusConfig::AudioEncoderOpusConfig()
+ : frame_size_ms(kDefaultFrameSizeMs),
+ num_channels(1),
+ application(ApplicationMode::kVoip),
+ bitrate_bps(32000),
+ fec_enabled(false),
+ cbr_enabled(false),
+ max_playback_rate_hz(48000),
+ complexity(kDefaultComplexity),
+ low_rate_complexity(kDefaultLowRateComplexity),
+ complexity_threshold_bps(12500),
+ complexity_threshold_window_bps(1500),
+ dtx_enabled(false),
+ uplink_bandwidth_update_interval_ms(200),
+ payload_type(-1) {}
+AudioEncoderOpusConfig::AudioEncoderOpusConfig(const AudioEncoderOpusConfig&) =
+ default;
+AudioEncoderOpusConfig::~AudioEncoderOpusConfig() = default;
+AudioEncoderOpusConfig& AudioEncoderOpusConfig::operator=(
+ const AudioEncoderOpusConfig&) = default;
+
+bool AudioEncoderOpusConfig::IsOk() const {
+ if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
+ return false;
+ if (num_channels != 1 && num_channels != 2)
+ return false;
+ if (!bitrate_bps)
+ return false;
+ if (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps)
+ return false;
+ if (complexity < 0 || complexity > 10)
+ return false;
+ if (low_rate_complexity < 0 || low_rate_complexity > 10)
+ return false;
+ return true;
+}
+} // namespace webrtc
diff --git a/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
new file mode 100644
index 0000000..bd25b54
--- /dev/null
+++ b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
+#define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "webrtc/base/optional.h"
+
+namespace webrtc {
+
+// NOTE: This struct is still under development and may change without notice.
+struct AudioEncoderOpusConfig {
+ static constexpr int kDefaultFrameSizeMs = 20;
+
+ // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
+ // bitrate should be in the range of 6000 to 510000, inclusive.
+ static constexpr int kMinBitrateBps = 6000;
+ static constexpr int kMaxBitrateBps = 510000;
+
+ AudioEncoderOpusConfig();
+ AudioEncoderOpusConfig(const AudioEncoderOpusConfig&);
+ ~AudioEncoderOpusConfig();
+ AudioEncoderOpusConfig& operator=(const AudioEncoderOpusConfig&);
+
+ bool IsOk() const; // Checks if the values are currently OK.
+
+ int frame_size_ms;
+ size_t num_channels;
+ enum class ApplicationMode { kVoip, kAudio };
+ ApplicationMode application;
+
+ // NOTE: This member must always be set.
+ // TODO(kwiberg): Turn it into just an int.
+ rtc::Optional<int> bitrate_bps;
+
+ bool fec_enabled;
+ bool cbr_enabled;
+ int max_playback_rate_hz;
+
+ // |complexity| is used when the bitrate goes above
+ // |complexity_threshold_bps| + |complexity_threshold_window_bps|;
+ // |low_rate_complexity| is used when the bitrate falls below
+ // |complexity_threshold_bps| - |complexity_threshold_window_bps|. In the
+ // interval in the middle, we keep using the most recent of the two
+ // complexity settings.
+ int complexity;
+ int low_rate_complexity;
+ int complexity_threshold_bps;
+ int complexity_threshold_window_bps;
+
+ bool dtx_enabled;
+ std::vector<int> supported_frame_lengths_ms;
+ int uplink_bandwidth_update_interval_ms;
+
+ // NOTE: This member isn't necessary, and will soon go away. See
+ // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+ int payload_type;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
diff --git a/webrtc/api/audio_codecs/test/BUILD.gn b/webrtc/api/audio_codecs/test/BUILD.gn
index 38ca736..685726f 100644
--- a/webrtc/api/audio_codecs/test/BUILD.gn
+++ b/webrtc/api/audio_codecs/test/BUILD.gn
@@ -21,6 +21,7 @@
]
deps = [
"..:audio_codecs_api",
+ "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed?
"../../../base:rtc_base_approved",
"../../../test:audio_codec_mocks",
"../../../test:test_support",
@@ -28,6 +29,7 @@
"../g722:audio_encoder_g722",
"../ilbc:audio_decoder_ilbc",
"../ilbc:audio_encoder_ilbc",
+ "../opus:audio_encoder_opus",
"//testing/gmock",
]
}
diff --git a/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc b/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
index c3c07c6..d08e7aa 100644
--- a/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
+++ b/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
@@ -11,6 +11,7 @@
#include "webrtc/api/audio_codecs/audio_encoder_factory_template.h"
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h"
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h"
+#include "webrtc/api/audio_codecs/opus/audio_encoder_opus.h"
#include "webrtc/base/ptr_util.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
@@ -149,4 +150,26 @@
EXPECT_EQ(8000, enc->SampleRateHz());
}
+TEST(AudioEncoderFactoryTemplateTest, Opus) {
+ auto factory = CreateAudioEncoderFactory<AudioEncoderOpus>();
+ AudioCodecInfo info = {48000, 1, 32000, 6000, 510000};
+ info.allow_comfort_noise = false;
+ info.supports_network_adaption = true;
+ EXPECT_THAT(
+ factory->GetSupportedEncoders(),
+ testing::ElementsAre(AudioCodecSpec{
+ {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
+ info}));
+ EXPECT_EQ(rtc::Optional<AudioCodecInfo>(),
+ factory->QueryAudioEncoder({"foo", 8000, 1}));
+ EXPECT_EQ(
+ rtc::Optional<AudioCodecInfo>(info),
+ factory->QueryAudioEncoder(
+ {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}));
+ EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"bar", 16000, 1}));
+ auto enc = factory->MakeAudioEncoder(17, {"opus", 48000, 2});
+ ASSERT_NE(nullptr, enc);
+ EXPECT_EQ(48000, enc->SampleRateHz());
+}
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 30460ba..1aa4031 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -829,6 +829,7 @@
":audio_network_adaptor",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
+ "../../api/audio_codecs/opus:audio_encoder_opus_config",
"../../base:protobuf_utils",
"../../base:rtc_base_approved",
"../../base:rtc_numerics",
@@ -840,11 +841,6 @@
]
defines = audio_codec_defines
- if (rtc_opus_variable_complexity) {
- defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
- } else {
- defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
- }
if (rtc_build_opus) {
public_deps += [ rtc_opus_dir ]
@@ -1485,6 +1481,7 @@
":neteq",
":neteq_tools",
"../../api/audio_codecs:audio_codecs_api",
+ "../../api/audio_codecs/opus:audio_encoder_opus",
"../../base:protobuf_utils",
"../../common_audio",
"../../test:test_main",
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 be32aef..9ebb63d 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -19,6 +19,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/numerics/exp_filter.h"
#include "webrtc/base/protobuf_utils.h"
+#include "webrtc/base/ptr_util.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/base/safe_minmax.h"
#include "webrtc/base/string_to_number.h"
@@ -48,11 +49,6 @@
constexpr int kOpusBitrateWbBps = 20000;
constexpr int kOpusBitrateFbBps = 32000;
-// Opus API allows a min bitrate of 500bps, but Opus documentation suggests
-// bitrate should be in the range of 6000 to 510000, inclusive.
-constexpr int kOpusMinBitrateBps = 6000;
-constexpr int kOpusMaxBitrateBps = 510000;
-
constexpr int kSampleRateHz = 48000;
constexpr int kDefaultMaxPlaybackRate = 48000;
@@ -133,8 +129,8 @@
return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
}
}();
- RTC_DCHECK_GE(bitrate, kOpusMinBitrateBps);
- RTC_DCHECK_LE(bitrate, kOpusMaxBitrateBps);
+ RTC_DCHECK_GE(bitrate, AudioEncoderOpusConfig::kMinBitrateBps);
+ RTC_DCHECK_LE(bitrate, AudioEncoderOpusConfig::kMaxBitrateBps);
return bitrate;
}
@@ -150,7 +146,8 @@
const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
if (bitrate) {
const int chosen_bitrate =
- std::max(kOpusMinBitrateBps, std::min(*bitrate, kOpusMaxBitrateBps));
+ std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+ std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
if (bitrate != chosen_bitrate) {
LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
<< " clamped to " << chosen_bitrate;
@@ -195,7 +192,7 @@
return *(std::end(kOpusSupportedFrameLengths) - 1);
}
- return AudioEncoderOpus::Config::kDefaultFrameSizeMs;
+ return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
}
void FindSupportedFrameLengths(int min_frame_length_ms,
@@ -211,8 +208,39 @@
RTC_DCHECK(std::is_sorted(out->begin(), out->end()));
}
+int GetBitrateBps(const AudioEncoderOpusConfig& config) {
+ RTC_DCHECK(config.IsOk());
+ return *config.bitrate_bps;
+}
+
} // namespace
+void AudioEncoderOpus::AppendSupportedEncoders(
+ std::vector<AudioCodecSpec>* specs) {
+ const SdpAudioFormat fmt = {
+ "opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}};
+ const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+ specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder(
+ const AudioEncoderOpusConfig& config) {
+ RTC_DCHECK(config.IsOk());
+ AudioCodecInfo info(48000, config.num_channels, *config.bitrate_bps,
+ AudioEncoderOpusConfig::kMinBitrateBps,
+ AudioEncoderOpusConfig::kMaxBitrateBps);
+ info.allow_comfort_noise = false;
+ info.supports_network_adaption = true;
+ return info;
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder(
+ const AudioEncoderOpusConfig& config,
+ int payload_type) {
+ RTC_DCHECK(config.IsOk());
+ return rtc::MakeUnique<AudioEncoderOpus>(config, payload_type);
+}
+
rtc::Optional<AudioCodecInfo> AudioEncoderOpus::QueryAudioEncoder(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
@@ -221,8 +249,9 @@
const int bitrate =
CalculateBitrate(GetMaxPlaybackRate(format), num_channels,
GetFormatParameter(format, "maxaveragebitrate"));
- AudioCodecInfo info(48000, num_channels, bitrate, kOpusMinBitrateBps,
- kOpusMaxBitrateBps);
+ AudioCodecInfo info(48000, num_channels, bitrate,
+ AudioEncoderOpusConfig::kMinBitrateBps,
+ AudioEncoderOpusConfig::kMaxBitrateBps);
info.allow_comfort_noise = false;
info.supports_network_adaption = true;
@@ -231,27 +260,36 @@
return rtc::Optional<AudioCodecInfo>();
}
-AudioEncoderOpus::Config AudioEncoderOpus::CreateConfig(
+AudioEncoderOpusConfig AudioEncoderOpus::CreateConfig(
+ int payload_type,
+ const SdpAudioFormat& format) {
+ auto opt_config = SdpToConfig(format);
+ RTC_CHECK(opt_config);
+ opt_config->payload_type = payload_type;
+ return *opt_config;
+}
+
+AudioEncoderOpusConfig AudioEncoderOpus::CreateConfig(
const CodecInst& codec_inst) {
- AudioEncoderOpus::Config config;
+ AudioEncoderOpusConfig config;
config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48);
config.num_channels = codec_inst.channels;
config.bitrate_bps = rtc::Optional<int>(codec_inst.rate);
- config.payload_type = codec_inst.pltype;
- config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
- : AudioEncoderOpus::kAudio;
+ config.application = config.num_channels == 1
+ ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+ : AudioEncoderOpusConfig::ApplicationMode::kAudio;
config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
-#if WEBRTC_OPUS_VARIABLE_COMPLEXITY
- config.low_rate_complexity = 9;
-#endif
return config;
}
-AudioEncoderOpus::Config AudioEncoderOpus::CreateConfig(
- int payload_type,
+rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
const SdpAudioFormat& format) {
- AudioEncoderOpus::Config config;
+ if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
+ format.clockrate_hz != 48000 || format.num_channels != 2) {
+ return rtc::Optional<AudioEncoderOpusConfig>();
+ }
+ AudioEncoderOpusConfig config;
config.num_channels = GetChannelCount(format);
config.frame_size_ms = GetFrameSizeMs(format);
config.max_playback_rate_hz = GetMaxPlaybackRate(format);
@@ -261,16 +299,14 @@
config.bitrate_bps = rtc::Optional<int>(
CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
GetFormatParameter(format, "maxaveragebitrate")));
- config.payload_type = payload_type;
- config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
- : AudioEncoderOpus::kAudio;
-#if WEBRTC_OPUS_VARIABLE_COMPLEXITY
- config.low_rate_complexity = 9;
-#endif
+ config.application = config.num_channels == 1
+ ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+ : AudioEncoderOpusConfig::ApplicationMode::kAudio;
constexpr int kMinANAFrameLength = kANASupportedFrameLengths[0];
constexpr int kMaxANAFrameLength =
kANASupportedFrameLengths[arraysize(kANASupportedFrameLengths) - 1];
+
// For now, minptime and maxptime are only used with ANA. If ptime is outside
// of this range, it will get adjusted once ANA takes hold. Ideally, we'd know
// if ANA was to be used when setting up the config, and adjust accordingly.
@@ -281,7 +317,25 @@
FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
&config.supported_frame_lengths_ms);
- return config;
+ RTC_DCHECK(config.IsOk());
+ return rtc::Optional<AudioEncoderOpusConfig>(config);
+}
+
+rtc::Optional<int> AudioEncoderOpus::GetNewComplexity(
+ const AudioEncoderOpusConfig& config) {
+ RTC_DCHECK(config.IsOk());
+ const int bitrate_bps = GetBitrateBps(config);
+ if (bitrate_bps >= config.complexity_threshold_bps -
+ config.complexity_threshold_window_bps &&
+ bitrate_bps <= config.complexity_threshold_bps +
+ config.complexity_threshold_window_bps) {
+ // Within the hysteresis window; make no change.
+ return rtc::Optional<int>();
+ } else {
+ return rtc::Optional<int>(bitrate_bps <= config.complexity_threshold_bps
+ ? config.low_rate_complexity
+ : config.complexity);
+ }
}
class AudioEncoderOpus::PacketLossFractionSmoother {
@@ -311,58 +365,16 @@
rtc::ExpFilter smoother_;
};
-AudioEncoderOpus::Config::Config() {
-#if WEBRTC_OPUS_VARIABLE_COMPLEXITY
- low_rate_complexity = 9;
-#endif
-}
-AudioEncoderOpus::Config::Config(const Config&) = default;
-AudioEncoderOpus::Config::~Config() = default;
-auto AudioEncoderOpus::Config::operator=(const Config&) -> Config& = default;
-
-bool AudioEncoderOpus::Config::IsOk() const {
- if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
- return false;
- if (num_channels != 1 && num_channels != 2)
- return false;
- if (bitrate_bps &&
- (*bitrate_bps < kOpusMinBitrateBps || *bitrate_bps > kOpusMaxBitrateBps))
- return false;
- if (complexity < 0 || complexity > 10)
- return false;
- if (low_rate_complexity < 0 || low_rate_complexity > 10)
- return false;
- return true;
-}
-
-int AudioEncoderOpus::Config::GetBitrateBps() const {
- RTC_DCHECK(IsOk());
- if (bitrate_bps)
- return *bitrate_bps; // Explicitly set value.
- else
- return num_channels == 1 ? 32000 : 64000; // Default value.
-}
-
-rtc::Optional<int> AudioEncoderOpus::Config::GetNewComplexity() const {
- RTC_DCHECK(IsOk());
- const int bitrate_bps = GetBitrateBps();
- if (bitrate_bps >=
- complexity_threshold_bps - complexity_threshold_window_bps &&
- bitrate_bps <=
- complexity_threshold_bps + complexity_threshold_window_bps) {
- // Within the hysteresis window; make no change.
- return rtc::Optional<int>();
- }
- return bitrate_bps <= complexity_threshold_bps
- ? rtc::Optional<int>(low_rate_complexity)
- : rtc::Optional<int>(complexity);
-}
+AudioEncoderOpus::AudioEncoderOpus(const AudioEncoderOpusConfig& config)
+ : AudioEncoderOpus(config, config.payload_type) {}
AudioEncoderOpus::AudioEncoderOpus(
- const Config& config,
+ const AudioEncoderOpusConfig& config,
+ int payload_type,
AudioNetworkAdaptorCreator&& audio_network_adaptor_creator,
std::unique_ptr<SmoothingFilter> bitrate_smoother)
- : send_side_bwe_with_overhead_(webrtc::field_trial::IsEnabled(
+ : payload_type_(payload_type),
+ send_side_bwe_with_overhead_(webrtc::field_trial::IsEnabled(
"WebRTC-SendSideBwe-WithOverhead")),
packet_loss_rate_(0.0),
inst_(nullptr),
@@ -379,15 +391,21 @@
? std::move(bitrate_smoother) : std::unique_ptr<SmoothingFilter>(
// We choose 5sec as initial time constant due to empirical data.
new SmoothingFilterImpl(5000))) {
+ RTC_DCHECK(0 <= payload_type && payload_type <= 127);
+
+ // Sanity check of the redundant payload type field that we want to get rid
+ // of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+ RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
+
RTC_CHECK(RecreateEncoderInstance(config));
}
AudioEncoderOpus::AudioEncoderOpus(const CodecInst& codec_inst)
- : AudioEncoderOpus(CreateConfig(codec_inst), nullptr) {}
+ : AudioEncoderOpus(CreateConfig(codec_inst), codec_inst.pltype) {}
AudioEncoderOpus::AudioEncoderOpus(int payload_type,
const SdpAudioFormat& format)
- : AudioEncoderOpus(CreateConfig(payload_type, format), nullptr) {}
+ : AudioEncoderOpus(*SdpToConfig(format), payload_type) {}
AudioEncoderOpus::~AudioEncoderOpus() {
RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
@@ -410,7 +428,7 @@
}
int AudioEncoderOpus::GetTargetBitrate() const {
- return config_.GetBitrateBps();
+ return GetBitrateBps(config_);
}
void AudioEncoderOpus::Reset() {
@@ -445,10 +463,10 @@
auto conf = config_;
switch (application) {
case Application::kSpeech:
- conf.application = AudioEncoderOpus::kVoip;
+ conf.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
break;
case Application::kAudio:
- conf.application = AudioEncoderOpus::kAudio;
+ conf.application = AudioEncoderOpusConfig::ApplicationMode::kAudio;
break;
}
return RecreateEncoderInstance(conf);
@@ -523,9 +541,10 @@
}
const int overhead_bps = static_cast<int>(
*overhead_bytes_per_packet_ * 8 * 100 / Num10MsFramesInNextPacket());
- SetTargetBitrate(std::min(
- kOpusMaxBitrateBps,
- std::max(kOpusMinBitrateBps, target_audio_bitrate_bps - overhead_bps)));
+ SetTargetBitrate(
+ std::min(AudioEncoderOpusConfig::kMaxBitrateBps,
+ std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+ target_audio_bitrate_bps - overhead_bps)));
} else {
SetTargetBitrate(target_audio_bitrate_bps);
}
@@ -597,7 +616,7 @@
config_.frame_size_ms = next_frame_length_ms_;
info.encoded_timestamp = first_timestamp_in_buffer_;
- info.payload_type = config_.payload_type;
+ info.payload_type = payload_type_;
info.send_even_if_empty = true; // Allows Opus to send empty packets.
info.speech = (info.encoded_bytes > 0);
info.encoder_type = CodecType::kOpus;
@@ -616,7 +635,7 @@
// Calculate the number of bytes we expect the encoder to produce,
// then multiply by two to give a wide margin for error.
const size_t bytes_per_millisecond =
- static_cast<size_t>(config_.GetBitrateBps() / (1000 * 8) + 1);
+ static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
const size_t approx_encoded_bytes =
Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
return 2 * approx_encoded_bytes;
@@ -625,7 +644,8 @@
// If the given config is OK, recreate the Opus encoder instance with those
// settings, save the config, and return true. Otherwise, do nothing and return
// false.
-bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
+bool AudioEncoderOpus::RecreateEncoderInstance(
+ const AudioEncoderOpusConfig& config) {
if (!config.IsOk())
return false;
config_ = config;
@@ -633,9 +653,13 @@
RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
input_buffer_.clear();
input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
- RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, config.num_channels,
- config.application));
- RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config.GetBitrateBps()));
+ RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(
+ &inst_, config.num_channels,
+ config.application ==
+ AudioEncoderOpusConfig::ApplicationMode::kVoip
+ ? 0
+ : 1));
+ RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, GetBitrateBps(config)));
if (config.fec_enabled) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
} else {
@@ -645,7 +669,7 @@
0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
// Use the default complexity if the start bitrate is within the hysteresis
// window.
- complexity_ = config.GetNewComplexity().value_or(config.complexity);
+ complexity_ = GetNewComplexity(config).value_or(config.complexity);
RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
if (config.dtx_enabled) {
RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
@@ -692,10 +716,11 @@
void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
config_.bitrate_bps = rtc::Optional<int>(rtc::SafeClamp<int>(
- bits_per_second, kOpusMinBitrateBps, kOpusMaxBitrateBps));
+ bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
+ AudioEncoderOpusConfig::kMaxBitrateBps));
RTC_DCHECK(config_.IsOk());
- RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.GetBitrateBps()));
- const auto new_complexity = config_.GetNewComplexity();
+ RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, GetBitrateBps(config_)));
+ const auto new_complexity = GetNewComplexity(config_);
if (new_complexity && complexity_ != *new_complexity) {
complexity_ = *new_complexity;
RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
@@ -728,11 +753,11 @@
AudioNetworkAdaptorImpl::Config config;
config.event_log = event_log;
return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
- config,
- ControllerManagerImpl::Create(
- config_string, NumChannels(), supported_frame_lengths_ms(),
- kOpusMinBitrateBps, num_channels_to_encode_, next_frame_length_ms_,
- GetTargetBitrate(), config_.fec_enabled, GetDtx())));
+ config, ControllerManagerImpl::Create(
+ config_string, NumChannels(), supported_frame_lengths_ms(),
+ AudioEncoderOpusConfig::kMinBitrateBps,
+ num_channels_to_encode_, next_frame_length_ms_,
+ GetTargetBitrate(), config_.fec_enabled, GetDtx())));
}
void AudioEncoderOpus::MaybeUpdateUplinkBandwidth() {
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index a1a4d70..e50441b 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -18,6 +18,7 @@
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
+#include "webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/protobuf_utils.h"
@@ -33,62 +34,42 @@
class AudioEncoderOpus final : public AudioEncoder {
public:
- enum ApplicationMode {
- kVoip = 0,
- kAudio = 1,
- };
+ static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+ static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+ static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+ const AudioEncoderOpusConfig&,
+ int payload_type);
- struct Config {
- Config();
- Config(const Config&);
- ~Config();
- Config& operator=(const Config&);
+ // NOTE: This alias will soon go away. See
+ // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+ using Config = AudioEncoderOpusConfig;
- bool IsOk() const;
- int GetBitrateBps() const;
- // 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
- // current bitrate is above or below complexity_threshold_bps.
- rtc::Optional<int> GetNewComplexity() const;
-
- static constexpr int kDefaultFrameSizeMs = 20;
- int frame_size_ms = kDefaultFrameSizeMs;
- size_t num_channels = 1;
- int payload_type = 120;
- ApplicationMode application = kVoip;
- rtc::Optional<int> bitrate_bps; // Unset means to use default value.
- bool fec_enabled = false;
- bool cbr_enabled = false;
- int max_playback_rate_hz = 48000;
- int complexity = kDefaultComplexity;
- // This value may change in the struct's constructor.
- int low_rate_complexity = kDefaultComplexity;
- // low_rate_complexity is used when the bitrate is below this threshold.
- int complexity_threshold_bps = 12500;
- int complexity_threshold_window_bps = 1500;
- bool dtx_enabled = false;
- std::vector<int> supported_frame_lengths_ms;
- int uplink_bandwidth_update_interval_ms = 200;
-
- private:
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
- // If we are on Android, iOS and/or ARM, use a lower complexity setting as
- // default, to save encoder complexity.
- static const int kDefaultComplexity = 5;
-#else
- static const int kDefaultComplexity = 9;
-#endif
- };
-
+ // NOTE: This function will soon go away. See
+ // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
static Config CreateConfig(int payload_type, const SdpAudioFormat& format);
- static Config CreateConfig(const CodecInst& codec_inst);
+
+ static AudioEncoderOpusConfig CreateConfig(const CodecInst& codec_inst);
+ static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
+ const SdpAudioFormat& format);
+
+ // 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
+ // current bitrate is above or below complexity_threshold_bps.
+ static rtc::Optional<int> GetNewComplexity(
+ const AudioEncoderOpusConfig& config);
using AudioNetworkAdaptorCreator =
std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&,
RtcEventLog*)>;
+
+ // NOTE: This constructor will soon go away. See
+ // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+ AudioEncoderOpus(const AudioEncoderOpusConfig& config);
+
AudioEncoderOpus(
- const Config& config,
+ const AudioEncoderOpusConfig& config,
+ int payload_type,
AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr,
std::unique_ptr<SmoothingFilter> bitrate_smoother = nullptr);
@@ -110,9 +91,9 @@
void Reset() override;
bool SetFec(bool enable) override;
- // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
- // being inactive. During that, it still sends 2 packets (one for content, one
- // for signaling) about every 400 ms.
+ // Set Opus DTX. Once enabled, Opus stops transmission, when it detects
+ // voice being inactive. During that, it still sends 2 packets (one for
+ // content, one for signaling) about every 400 ms.
bool SetDtx(bool enable) override;
bool GetDtx() const override;
@@ -138,7 +119,9 @@
// Getters for testing.
float packet_loss_rate() const { return packet_loss_rate_; }
- ApplicationMode application() const { return config_.application; }
+ AudioEncoderOpusConfig::ApplicationMode application() const {
+ return config_.application;
+ }
bool fec_enabled() const { return config_.fec_enabled; }
size_t num_channels_to_encode() const { return num_channels_to_encode_; }
int next_frame_length_ms() const { return next_frame_length_ms_; }
@@ -154,7 +137,7 @@
size_t Num10msFramesPerPacket() const;
size_t SamplesPer10msFrame() const;
size_t SufficientOutputBufferSize() const;
- bool RecreateEncoderInstance(const Config& config);
+ bool RecreateEncoderInstance(const AudioEncoderOpusConfig& config);
void SetFrameLength(int frame_length_ms);
void SetNumChannelsToEncode(size_t num_channels_to_encode);
void SetProjectedPacketLossRate(float fraction);
@@ -170,7 +153,8 @@
void MaybeUpdateUplinkBandwidth();
- Config config_;
+ AudioEncoderOpusConfig config_;
+ const int payload_type_;
const bool send_side_bwe_with_overhead_;
float packet_loss_rate_;
std::vector<int16_t> input_buffer_;
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 76574d6..ecaea5d 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
@@ -33,22 +33,22 @@
const CodecInst kDefaultOpusSettings = {105, "opus", 48000, 960, 1, 32000};
constexpr int64_t kInitialTimeUs = 12345678;
-AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
- AudioEncoderOpus::Config config;
+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 = rtc::Optional<int>(codec_inst.rate);
- config.payload_type = codec_inst.pltype;
- config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
- : AudioEncoderOpus::kAudio;
+ 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;
}
-AudioEncoderOpus::Config CreateConfigWithParameters(
+AudioEncoderOpusConfig CreateConfigWithParameters(
const SdpAudioFormat::Parameters& params) {
- SdpAudioFormat format("opus", 48000, 2, params);
- return AudioEncoderOpus::CreateConfig(0, format);
+ const SdpAudioFormat format("opus", 48000, 2, params);
+ return *AudioEncoderOpus::SdpToConfig(format);
}
struct AudioEncoderOpusStates {
@@ -56,7 +56,7 @@
MockSmoothingFilter* mock_bitrate_smoother;
std::unique_ptr<AudioEncoderOpus> encoder;
std::unique_ptr<rtc::ScopedFakeClock> fake_clock;
- AudioEncoderOpus::Config config;
+ AudioEncoderOpusConfig config;
};
AudioEncoderOpusStates CreateCodec(size_t num_channels) {
@@ -87,7 +87,8 @@
new MockSmoothingFilter());
states.mock_bitrate_smoother = bitrate_smoother.get();
- states.encoder.reset(new AudioEncoderOpus(states.config, std::move(creator),
+ states.encoder.reset(new AudioEncoderOpus(states.config, codec_inst.pltype,
+ std::move(creator),
std::move(bitrate_smoother)));
return states;
}
@@ -142,19 +143,22 @@
TEST(AudioEncoderOpusTest, DefaultApplicationModeMono) {
auto states = CreateCodec(1);
- EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
+ EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+ states.encoder->application());
}
TEST(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
auto states = CreateCodec(2);
- EXPECT_EQ(AudioEncoderOpus::kAudio, states.encoder->application());
+ EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
+ states.encoder->application());
}
TEST(AudioEncoderOpusTest, ChangeApplicationMode) {
auto states = CreateCodec(2);
EXPECT_TRUE(
states.encoder->SetApplication(AudioEncoder::Application::kSpeech));
- EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
+ EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+ states.encoder->application());
}
TEST(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
@@ -163,17 +167,20 @@
// Trigger a reset.
states.encoder->Reset();
// Verify that the mode is still kAudio.
- EXPECT_EQ(AudioEncoderOpus::kAudio, states.encoder->application());
+ EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
+ states.encoder->application());
// Now change to kVoip.
EXPECT_TRUE(
states.encoder->SetApplication(AudioEncoder::Application::kSpeech));
- EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
+ EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+ states.encoder->application());
// Trigger a reset again.
states.encoder->Reset();
// Verify that the mode is still kVoip.
- EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
+ EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+ states.encoder->application());
}
TEST(AudioEncoderOpusTest, ToggleDtx) {
@@ -452,25 +459,25 @@
// Verifies that the complexity adaptation in the config works as intended.
TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
- AudioEncoderOpus::Config config;
+ AudioEncoderOpusConfig config;
config.low_rate_complexity = 8;
config.complexity = 6;
// Bitrate within hysteresis window. Expect empty output.
config.bitrate_bps = rtc::Optional<int>(12500);
- EXPECT_EQ(rtc::Optional<int>(), config.GetNewComplexity());
+ EXPECT_EQ(rtc::Optional<int>(), AudioEncoderOpus::GetNewComplexity(config));
// Bitrate below hysteresis window. Expect higher complexity.
config.bitrate_bps = rtc::Optional<int>(10999);
- EXPECT_EQ(rtc::Optional<int>(8), config.GetNewComplexity());
+ EXPECT_EQ(rtc::Optional<int>(8), AudioEncoderOpus::GetNewComplexity(config));
// Bitrate within hysteresis window. Expect empty output.
config.bitrate_bps = rtc::Optional<int>(12500);
- EXPECT_EQ(rtc::Optional<int>(), config.GetNewComplexity());
+ EXPECT_EQ(rtc::Optional<int>(), AudioEncoderOpus::GetNewComplexity(config));
// Bitrate above hysteresis window. Expect lower complexity.
config.bitrate_bps = rtc::Optional<int>(14001);
- EXPECT_EQ(rtc::Optional<int>(6), config.GetNewComplexity());
+ EXPECT_EQ(rtc::Optional<int>(6), AudioEncoderOpus::GetNewComplexity(config));
}
TEST(AudioEncoderOpusTest, EmptyConfigDoesNotAffectEncoderSettings) {
@@ -552,84 +559,82 @@
}
TEST(AudioEncoderOpusTest, TestConfigDefaults) {
- const AudioEncoderOpus::Config config =
- AudioEncoderOpus::CreateConfig(0, {"opus", 48000, 2});
-
- EXPECT_EQ(48000, config.max_playback_rate_hz);
- EXPECT_EQ(1u, config.num_channels);
- EXPECT_FALSE(config.fec_enabled);
- EXPECT_FALSE(config.dtx_enabled);
- EXPECT_EQ(20, config.frame_size_ms);
+ const auto config_opt = AudioEncoderOpus::SdpToConfig({"opus", 48000, 2});
+ ASSERT_TRUE(config_opt);
+ EXPECT_EQ(48000, config_opt->max_playback_rate_hz);
+ EXPECT_EQ(1u, config_opt->num_channels);
+ EXPECT_FALSE(config_opt->fec_enabled);
+ EXPECT_FALSE(config_opt->dtx_enabled);
+ EXPECT_EQ(20, config_opt->frame_size_ms);
}
TEST(AudioEncoderOpusTest, TestConfigFromParams) {
- AudioEncoderOpus::Config config;
+ const auto config1 = CreateConfigWithParameters({{"stereo", "0"}});
+ EXPECT_EQ(1U, config1.num_channels);
- config = CreateConfigWithParameters({{"stereo", "0"}});
- EXPECT_EQ(1U, config.num_channels);
+ const auto config2 = CreateConfigWithParameters({{"stereo", "1"}});
+ EXPECT_EQ(2U, config2.num_channels);
- config = CreateConfigWithParameters({{"stereo", "1"}});
- EXPECT_EQ(2U, config.num_channels);
+ const auto config3 = CreateConfigWithParameters({{"useinbandfec", "0"}});
+ EXPECT_FALSE(config3.fec_enabled);
- config = CreateConfigWithParameters({{"useinbandfec", "0"}});
- EXPECT_FALSE(config.fec_enabled);
+ const auto config4 = CreateConfigWithParameters({{"useinbandfec", "1"}});
+ EXPECT_TRUE(config4.fec_enabled);
- config = CreateConfigWithParameters({{"useinbandfec", "1"}});
- EXPECT_TRUE(config.fec_enabled);
+ const auto config5 = CreateConfigWithParameters({{"usedtx", "0"}});
+ EXPECT_FALSE(config5.dtx_enabled);
- config = CreateConfigWithParameters({{"usedtx", "0"}});
- EXPECT_FALSE(config.dtx_enabled);
+ const auto config6 = CreateConfigWithParameters({{"usedtx", "1"}});
+ EXPECT_TRUE(config6.dtx_enabled);
- config = CreateConfigWithParameters({{"usedtx", "1"}});
- EXPECT_TRUE(config.dtx_enabled);
+ const auto config7 = CreateConfigWithParameters({{"cbr", "0"}});
+ EXPECT_FALSE(config7.cbr_enabled);
- config = CreateConfigWithParameters({{"cbr", "0"}});
- EXPECT_FALSE(config.cbr_enabled);
+ const auto config8 = CreateConfigWithParameters({{"cbr", "1"}});
+ EXPECT_TRUE(config8.cbr_enabled);
- config = CreateConfigWithParameters({{"cbr", "1"}});
- EXPECT_TRUE(config.cbr_enabled);
+ const auto config9 =
+ CreateConfigWithParameters({{"maxplaybackrate", "12345"}});
+ EXPECT_EQ(12345, config9.max_playback_rate_hz);
- config = CreateConfigWithParameters({{"maxplaybackrate", "12345"}});
- EXPECT_EQ(12345, config.max_playback_rate_hz);
+ const auto config10 =
+ CreateConfigWithParameters({{"maxaveragebitrate", "96000"}});
+ EXPECT_EQ(96000, config10.bitrate_bps);
- config = CreateConfigWithParameters({{"maxaveragebitrate", "96000"}});
- EXPECT_EQ(96000, config.bitrate_bps);
-
- config = CreateConfigWithParameters({{"maxptime", "40"}});
- for (int frame_length : config.supported_frame_lengths_ms) {
+ const auto config11 = CreateConfigWithParameters({{"maxptime", "40"}});
+ for (int frame_length : config11.supported_frame_lengths_ms) {
EXPECT_LE(frame_length, 40);
}
- config = CreateConfigWithParameters({{"minptime", "40"}});
- for (int frame_length : config.supported_frame_lengths_ms) {
+ const auto config12 = CreateConfigWithParameters({{"minptime", "40"}});
+ for (int frame_length : config12.supported_frame_lengths_ms) {
EXPECT_GE(frame_length, 40);
}
- config = CreateConfigWithParameters({{"ptime", "40"}});
- EXPECT_EQ(40, config.frame_size_ms);
+ const auto config13 = CreateConfigWithParameters({{"ptime", "40"}});
+ EXPECT_EQ(40, config13.frame_size_ms);
constexpr int kMinSupportedFrameLength = 10;
constexpr int kMaxSupportedFrameLength =
WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
- config = CreateConfigWithParameters({{"ptime", "1"}});
- EXPECT_EQ(kMinSupportedFrameLength, config.frame_size_ms);
+ const auto config14 = CreateConfigWithParameters({{"ptime", "1"}});
+ EXPECT_EQ(kMinSupportedFrameLength, config14.frame_size_ms);
- config = CreateConfigWithParameters({{"ptime", "2000"}});
- EXPECT_EQ(kMaxSupportedFrameLength, config.frame_size_ms);
+ const auto config15 = CreateConfigWithParameters({{"ptime", "2000"}});
+ EXPECT_EQ(kMaxSupportedFrameLength, config15.frame_size_ms);
}
TEST(AudioEncoderOpusTest, TestConfigFromInvalidParams) {
const webrtc::SdpAudioFormat format("opus", 48000, 2);
- const AudioEncoderOpus::Config default_config =
- AudioEncoderOpus::CreateConfig(0, format);
+ const auto default_config = *AudioEncoderOpus::SdpToConfig(format);
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
const std::vector<int> default_supported_frame_lengths_ms({20, 60, 120});
#else
const std::vector<int> default_supported_frame_lengths_ms({20, 60});
#endif
- AudioEncoderOpus::Config config;
+ AudioEncoderOpusConfig config;
config = CreateConfigWithParameters({{"stereo", "invalid"}});
EXPECT_EQ(default_config.num_channels, config.num_channels);
@@ -681,18 +686,18 @@
// range of 6000 and 510000
TEST(AudioEncoderOpusTest, SetSendCodecOpusMaxAverageBitrate) {
// Ignore if less than 6000.
- const AudioEncoderOpus::Config config1 = AudioEncoderOpus::CreateConfig(
- 0, {"opus", 48000, 2, {{"maxaveragebitrate", "5999"}}});
- EXPECT_EQ(6000, config1.bitrate_bps);
+ const auto config1 = AudioEncoderOpus::SdpToConfig(
+ {"opus", 48000, 2, {{"maxaveragebitrate", "5999"}}});
+ EXPECT_EQ(6000, config1->bitrate_bps);
// Ignore if larger than 510000.
- const AudioEncoderOpus::Config config2 = AudioEncoderOpus::CreateConfig(
- 0, {"opus", 48000, 2, {{"maxaveragebitrate", "510001"}}});
- EXPECT_EQ(510000, config2.bitrate_bps);
+ const auto config2 = AudioEncoderOpus::SdpToConfig(
+ {"opus", 48000, 2, {{"maxaveragebitrate", "510001"}}});
+ EXPECT_EQ(510000, config2->bitrate_bps);
- const AudioEncoderOpus::Config config3 = AudioEncoderOpus::CreateConfig(
- 0, {"opus", 48000, 2, {{"maxaveragebitrate", "200000"}}});
- EXPECT_EQ(200000, config3.bitrate_bps);
+ const auto config3 = AudioEncoderOpus::SdpToConfig(
+ {"opus", 48000, 2, {{"maxaveragebitrate", "200000"}}});
+ EXPECT_EQ(200000, config3->bitrate_bps);
}
// Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
index b199912..8b8f9a1 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
@@ -19,9 +19,10 @@
namespace webrtc {
namespace {
-int64_t RunComplexityTest(const AudioEncoderOpus::Config& config) {
+int64_t RunComplexityTest(const AudioEncoderOpusConfig& config) {
// Create encoder.
- AudioEncoderOpus encoder(config);
+ constexpr int payload_type = 17;
+ AudioEncoderOpus encoder(config, payload_type);
// Open speech file.
const std::string kInputFileName =
webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
@@ -60,7 +61,7 @@
// the lower rate.
TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) {
// Create config.
- AudioEncoderOpus::Config config;
+ AudioEncoderOpusConfig config;
// The limit -- including the hysteresis window -- at which the complexity
// shuold be increased.
config.bitrate_bps = rtc::Optional<int>(11000 - 1);
@@ -80,7 +81,7 @@
// that the resulting ratio is less than 100% at all times.
TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) {
// Create config.
- AudioEncoderOpus::Config config;
+ AudioEncoderOpusConfig config;
// The limit -- including the hysteresis window -- at which the complexity
// shuold be increased (but not in this test since complexity adaptation is
// disabled).
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 15a89e9..1294f23 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -17,6 +17,7 @@
#include <string>
#include <vector>
+#include "webrtc/api/audio_codecs/opus/audio_encoder_opus.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
@@ -28,7 +29,6 @@
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
-#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
@@ -433,11 +433,10 @@
frame_size_ = 480;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderOpus(1);
- AudioEncoderOpus::Config config;
+ AudioEncoderOpusConfig config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
- config.payload_type = payload_type_;
- config.application = AudioEncoderOpus::kVoip;
- audio_encoder_.reset(new AudioEncoderOpus(config));
+ config.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
+ audio_encoder_ = AudioEncoderOpus::MakeAudioEncoder(config, payload_type_);
}
};
@@ -447,12 +446,11 @@
channels_ = 2;
delete decoder_;
decoder_ = new AudioDecoderOpus(2);
- AudioEncoderOpus::Config config;
+ AudioEncoderOpusConfig config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
config.num_channels = 2;
- config.payload_type = payload_type_;
- config.application = AudioEncoderOpus::kAudio;
- audio_encoder_.reset(new AudioEncoderOpus(config));
+ config.application = AudioEncoderOpusConfig::ApplicationMode::kAudio;
+ audio_encoder_ = AudioEncoderOpus::MakeAudioEncoder(config, payload_type_);
}
};
