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New interface: AudioEncoderMutable

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With implementations for all codecs. It has no users yet. This new
interface is the same as AudioEncoder (in fact it is a subclass) but
it allows changing some parameters after construction.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
R=jmarusic@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51679004

Cr-Commit-Position: refs/heads/master@{#9149}
This commit is contained in:
Karl Wiberg
2015-05-07 12:35:12 +02:00
parent 81ea54eaac
commit dcccab3ebb
21 changed files with 784 additions and 27 deletions
Download Patch File Download Diff File Expand all files Collapse all files

22
webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
View File

@ -12,6 +12,7 @@
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
namespace webrtc {
@ -22,6 +23,10 @@ const int kSampleRateHz = 16000;
} // namespace
bool AudioEncoderG722::Config::IsOk() const {
return (frame_size_ms % 10 == 0) && (num_channels >= 1);
}
AudioEncoderG722::EncoderState::EncoderState() {
CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
CHECK_EQ(0, WebRtcG722_EncoderInit(encoder));
@ -39,8 +44,7 @@ AudioEncoderG722::AudioEncoderG722(const Config& config)
first_timestamp_in_buffer_(0),
encoders_(new EncoderState[num_channels_]),
interleave_buffer_(2 * num_channels_) {
CHECK_EQ(config.frame_size_ms % 10, 0)
<< "Frame size must be an integer multiple of 10 ms.";
CHECK(config.IsOk());
const int samples_per_channel =
kSampleRateHz / 100 * num_10ms_frames_per_packet_;
for (int i = 0; i < num_channels_; ++i) {
@ -134,4 +138,18 @@ int AudioEncoderG722::SamplesPerChannel() const {
return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
}
namespace {
AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
AudioEncoderG722::Config config;
config.num_channels = codec_inst.channels;
config.frame_size_ms = codec_inst.pacsize / 16;
config.payload_type = codec_inst.pltype;
return config;
}
} // namespace
AudioEncoderMutableG722::AudioEncoderMutableG722(const CodecInst& codec_inst)
: AudioEncoderMutableImpl<AudioEncoderG722>(CreateConfig(codec_inst)) {
}
} // namespace webrtc

12
webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
View File

@ -14,6 +14,7 @@
#include "webrtc/base/buffer.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
namespace webrtc {
@ -22,6 +23,7 @@ class AudioEncoderG722 : public AudioEncoder {
public:
struct Config {
Config() : payload_type(9), frame_size_ms(20), num_channels(1) {}
bool IsOk() const;
int payload_type;
int frame_size_ms;
@ -37,8 +39,6 @@ class AudioEncoderG722 : public AudioEncoder {
int RtpTimestampRateHz() const override;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
protected:
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
@ -65,5 +65,13 @@ class AudioEncoderG722 : public AudioEncoder {
rtc::Buffer interleave_buffer_;
};
struct CodecInst;
class AudioEncoderMutableG722
: public AudioEncoderMutableImpl<AudioEncoderG722> {
public:
explicit AudioEncoderMutableG722(const CodecInst& codec_inst);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_