Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 126 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
new file mode 100644
index 0000000000..65e2da479d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -0,0 +1,126 @@
+/*
+ *  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 "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+bool AudioEncoderPcm::Config::IsOk() const {
+  return (frame_size_ms % 10 == 0) && (num_channels >= 1);
+}
+
+AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
+    : sample_rate_hz_(sample_rate_hz),
+      num_channels_(config.num_channels),
+      payload_type_(config.payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      full_frame_samples_(config.num_channels * config.frame_size_ms *
+                          sample_rate_hz / 1000),
+      first_timestamp_in_buffer_(0) {
+  RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
+  RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
+      << "Frame size must be an integer multiple of 10 ms.";
+  speech_buffer_.reserve(full_frame_samples_);
+}
+
+AudioEncoderPcm::~AudioEncoderPcm() = default;
+
+int AudioEncoderPcm::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioEncoderPcm::NumChannels() const {
+  return num_channels_;
+}
+
+size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderPcm::GetTargetBitrate() const {
+  return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
+                          NumChannels());
+}
+
+AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (speech_buffer_.empty()) {
+    first_timestamp_in_buffer_ = rtp_timestamp;
+  }
+  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
+  if (speech_buffer_.size() < full_frame_samples_) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
+  EncodedInfo info;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoded_bytes = encoded->AppendData(
+      full_frame_samples_ * BytesPerSample(),
+      [&](rtc::ArrayView<uint8_t> encoded) {
+        return EncodeCall(&speech_buffer_[0], full_frame_samples_,
+                          encoded.data());
+      });
+  speech_buffer_.clear();
+  info.encoder_type = GetCodecType();
+  return info;
+}
+
+void AudioEncoderPcm::Reset() {
+  speech_buffer_.clear();
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderPcm::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeA(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmA::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmA;
+}
+
+size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeU(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmU::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmU;
+}
+
+}  // namespace webrtc