webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
Mirko Bonadei 682aac5103 Enable clang::find_bad_constructs for audio_coding (part 1/2).
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7
Reviewed-on: https://webrtc-review.googlesource.com/89743
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24053}
2018-07-20 13:07:47 +00:00

69 lines
2.2 KiB
C++

/*
* Copyright (c) 2016 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.
*/
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
#include <memory>
#include "api/audio_codecs/audio_encoder.h"
#include "modules/audio_coding/neteq/tools/neteq_input.h"
namespace webrtc {
namespace test {
// This class provides a NetEqInput that takes audio from a generator object and
// encodes it using a given audio encoder.
class EncodeNetEqInput : public NetEqInput {
public:
// Generator class, to be provided to the EncodeNetEqInput constructor.
class Generator {
public:
virtual ~Generator() = default;
// Returns the next num_samples values from the signal generator.
virtual rtc::ArrayView<const int16_t> Generate(size_t num_samples) = 0;
};
// The source will end after the given input duration.
EncodeNetEqInput(std::unique_ptr<Generator> generator,
std::unique_ptr<AudioEncoder> encoder,
int64_t input_duration_ms);
~EncodeNetEqInput() override;
absl::optional<int64_t> NextPacketTime() const override;
absl::optional<int64_t> NextOutputEventTime() const override;
std::unique_ptr<PacketData> PopPacket() override;
void AdvanceOutputEvent() override;
bool ended() const override;
absl::optional<RTPHeader> NextHeader() const override;
private:
static constexpr int64_t kOutputPeriodMs = 10;
void CreatePacket();
std::unique_ptr<Generator> generator_;
std::unique_ptr<AudioEncoder> encoder_;
std::unique_ptr<PacketData> packet_data_;
uint32_t rtp_timestamp_ = 0;
int16_t sequence_number_ = 0;
int64_t next_packet_time_ms_ = 0;
int64_t next_output_event_ms_ = 0;
const int64_t input_duration_ms_;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_