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-rw-r--r--third_party/libwebrtc/api/audio_codecs/audio_encoder.cc114
1 files changed, 114 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc b/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc
new file mode 100644
index 0000000000..31bb8739f7
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc
@@ -0,0 +1,114 @@
+/*
+ * 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 "api/audio_codecs/audio_encoder.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+ANAStats::ANAStats() = default;
+ANAStats::~ANAStats() = default;
+ANAStats::ANAStats(const ANAStats&) = default;
+
+AudioEncoder::EncodedInfo::EncodedInfo() = default;
+AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
+AudioEncoder::EncodedInfo::~EncodedInfo() = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
+ const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
+ default;
+
+int AudioEncoder::RtpTimestampRateHz() const {
+ return SampleRateHz();
+}
+
+AudioEncoder::EncodedInfo AudioEncoder::Encode(
+ uint32_t rtp_timestamp,
+ rtc::ArrayView<const int16_t> audio,
+ rtc::Buffer* encoded) {
+ TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
+ RTC_CHECK_EQ(audio.size(),
+ static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
+
+ const size_t old_size = encoded->size();
+ EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
+ RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
+ return info;
+}
+
+bool AudioEncoder::SetFec(bool enable) {
+ return !enable;
+}
+
+bool AudioEncoder::SetDtx(bool enable) {
+ return !enable;
+}
+
+bool AudioEncoder::GetDtx() const {
+ return false;
+}
+
+bool AudioEncoder::SetApplication(Application application) {
+ return false;
+}
+
+void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
+
+void AudioEncoder::SetTargetBitrate(int target_bps) {}
+
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoder::ReclaimContainedEncoders() {
+ return nullptr;
+}
+
+bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
+ RtcEventLog* event_log) {
+ return false;
+}
+
+void AudioEncoder::DisableAudioNetworkAdaptor() {}
+
+void AudioEncoder::OnReceivedUplinkPacketLossFraction(
+ float uplink_packet_loss_fraction) {}
+
+void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) {
+ RTC_DCHECK_NOTREACHED();
+}
+
+void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
+ OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
+}
+
+void AudioEncoder::OnReceivedUplinkBandwidth(
+ int target_audio_bitrate_bps,
+ absl::optional<int64_t> bwe_period_ms) {}
+
+void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
+ OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
+ update.bwe_period.ms());
+}
+
+void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
+
+void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
+
+void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
+ int max_frame_length_ms) {}
+
+ANAStats AudioEncoder::GetANAStats() const {
+ return ANAStats();
+}
+
+constexpr int AudioEncoder::kMaxNumberOfChannels;
+} // namespace webrtc