8b3db59b6e
This reverts commit 5341aaccdb64e3336abf5875e8828222446adffa. Reason for revert: Order of initialization of global static strings. Original change's description: > Reland of https://webrtc-review.googlesource.com/c/src/+/114883 > > The difference to the original is new bitexactness strings AND > global static file string constants. The reason for reland is breaking > downstream projects. > > Original CL description: > > Tests for multi-stream Opus. > > This CL (mainly) adds bit-exactness tests for multi-stream Opus. The > tests are in audio_coding_unittest.cc. Some refactoring of > AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it > possible. A few checks for "channels \in {1, 2}" are replaced with > "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few > other changes are made to be able to write and read multi-channel WAV > files. > > The SDP changes are NOT included; as of this CL there is no way to set > up a multi-stream opus en/de-coder from SDP strings. > > Bug: webrtc:8649 > Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2 > Reviewed-on: https://webrtc-review.googlesource.com/c/123387 > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26774} TBR=aleloi@webrtc.org,ossu@webrtc.org Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8649 Reviewed-on: https://webrtc-review.googlesource.com/c/123580 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26777}
71 lines
2.3 KiB
C++
71 lines
2.3 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
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namespace webrtc {
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namespace {
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#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
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// If we are on Android, iOS and/or ARM, use a lower complexity setting by
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// default, to save encoder complexity.
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constexpr int kDefaultComplexity = 5;
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#else
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constexpr int kDefaultComplexity = 9;
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#endif
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constexpr int kDefaultLowRateComplexity =
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WEBRTC_OPUS_VARIABLE_COMPLEXITY ? 9 : kDefaultComplexity;
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} // namespace
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constexpr int AudioEncoderOpusConfig::kDefaultFrameSizeMs;
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constexpr int AudioEncoderOpusConfig::kMinBitrateBps;
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constexpr int AudioEncoderOpusConfig::kMaxBitrateBps;
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AudioEncoderOpusConfig::AudioEncoderOpusConfig()
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: frame_size_ms(kDefaultFrameSizeMs),
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num_channels(1),
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application(ApplicationMode::kVoip),
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bitrate_bps(32000),
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fec_enabled(false),
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cbr_enabled(false),
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max_playback_rate_hz(48000),
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complexity(kDefaultComplexity),
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low_rate_complexity(kDefaultLowRateComplexity),
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complexity_threshold_bps(12500),
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complexity_threshold_window_bps(1500),
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dtx_enabled(false),
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uplink_bandwidth_update_interval_ms(200),
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payload_type(-1) {}
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AudioEncoderOpusConfig::AudioEncoderOpusConfig(const AudioEncoderOpusConfig&) =
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default;
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AudioEncoderOpusConfig::~AudioEncoderOpusConfig() = default;
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AudioEncoderOpusConfig& AudioEncoderOpusConfig::operator=(
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const AudioEncoderOpusConfig&) = default;
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bool AudioEncoderOpusConfig::IsOk() const {
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if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
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return false;
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if (num_channels != 1 && num_channels != 2)
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return false;
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if (!bitrate_bps)
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return false;
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if (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps)
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return false;
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if (complexity < 0 || complexity > 10)
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return false;
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if (low_rate_complexity < 0 || low_rate_complexity > 10)
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return false;
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return true;
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}
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} // namespace webrtc
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