Index: webrtc/api/audio_codecs/opus/audio_encoder_opus.cc |
diff --git a/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc b/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc |
deleted file mode 100644 |
index 0969561ed14f8afca21106d9aaaea19ac60fc74f..0000000000000000000000000000000000000000 |
--- a/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc |
+++ /dev/null |
@@ -1,52 +0,0 @@ |
-/* |
- * 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.h" |
- |
-#include <memory> |
-#include <vector> |
- |
-#include "webrtc/base/ptr_util.h" |
-#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h" |
- |
-namespace webrtc { |
- |
-rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig( |
- const SdpAudioFormat& format) { |
- return AudioEncoderOpusImpl::SdpToConfig(format); |
-} |
- |
-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<AudioEncoderOpusImpl>(config, payload_type); |
-} |
- |
-} // namespace webrtc |