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Diffstat (limited to 'webrtc/modules/audio_coding/codecs/audio_encoder.cc')
-rw-r--r--webrtc/modules/audio_coding/codecs/audio_encoder.cc55
1 files changed, 0 insertions, 55 deletions
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
deleted file mode 100644
index 6d76300..0000000
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Copyright (c) 2014 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/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/base/checks.h"
-
-namespace webrtc {
-
-AudioEncoder::EncodedInfo::EncodedInfo() = default;
-
-AudioEncoder::EncodedInfo::~EncodedInfo() = default;
-
-int AudioEncoder::RtpTimestampRateHz() const {
- return SampleRateHz();
-}
-
-AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
- const int16_t* audio,
- size_t num_samples_per_channel,
- size_t max_encoded_bytes,
- uint8_t* encoded) {
- RTC_CHECK_EQ(num_samples_per_channel,
- static_cast<size_t>(SampleRateHz() / 100));
- EncodedInfo info =
- EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
- RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
- return info;
-}
-
-bool AudioEncoder::SetFec(bool enable) {
- return !enable;
-}
-
-bool AudioEncoder::SetDtx(bool enable) {
- return !enable;
-}
-
-bool AudioEncoder::SetApplication(Application application) {
- return false;
-}
-
-void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
-
-void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
-
-void AudioEncoder::SetTargetBitrate(int target_bps) {}
-
-} // namespace webrtc