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Let Add10MsData method do the encoding work as well

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This change essentially makes the Process method a no-op. All it does
now is to return a stored value from the last encoding.

The purpose of this change is to forge the Add... and Process methods
into one and the same.

BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38229004

Cr-Commit-Position: refs/heads/master@{#8499}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8499 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2015-02-25 10:33:10 +00:00
parent 4aef5fef18
commit af82f75690
2 changed files with 24 additions and 4 deletions
Download Patch File Download Diff File Expand all files Collapse all files

24
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
View File

@ -145,10 +145,10 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
aux_rtp_header_(NULL),
receiver_initialized_(false),
first_10ms_data_(false),
last_encode_value_(0),
callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
packetization_callback_(NULL),
vad_callback_(NULL) {
// Nullify send codec memory, set payload type and set codec name to
// invalid values.
const char no_name[] = "noCodecRegistered";
@ -231,8 +231,12 @@ int64_t AudioCodingModuleImpl::TimeUntilNextProcess() {
// (send_codec_inst_.plfreq / 1000);
}
// Process any pending tasks such as timeouts.
int32_t AudioCodingModuleImpl::Process() {
CriticalSectionScoped lock(acm_crit_sect_);
return last_encode_value_;
}
int32_t AudioCodingModuleImpl::Encode() {
// Make room for 1 RED payload.
uint8_t stream[2 * MAX_PAYLOAD_SIZE_BYTE];
// TODO(turajs): |length_bytes| & |red_length_bytes| can be of type int if
@ -758,8 +762,20 @@ int AudioCodingModuleImpl::RegisterTransportCallback(
}
// Add 10MS of raw (PCM) audio data to the encoder.
int AudioCodingModuleImpl::Add10MsData(
const AudioFrame& audio_frame) {
int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
int r = Add10MsDataInternal(audio_frame);
if (r < 0) {
CriticalSectionScoped lock(acm_crit_sect_);
last_encode_value_ = -1;
} else {
int r_encode = Encode();
CriticalSectionScoped lock(acm_crit_sect_);
last_encode_value_ = r_encode;
}
return r;
}
int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame) {
if (audio_frame.samples_per_channel_ <= 0) {
assert(false);
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,

4
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
View File

@ -240,6 +240,9 @@ class AudioCodingModuleImpl : public AudioCodingModule {
AudioDecodingCallStats* stats) const OVERRIDE;
private:
int Add10MsDataInternal(const AudioFrame& audio_frame);
int Encode();
ACMGenericCodec* CreateCodec(const CodecInst& codec);
int InitializeReceiverSafe() EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
@ -336,6 +339,7 @@ class AudioCodingModuleImpl : public AudioCodingModule {
AudioFrame preprocess_frame_ GUARDED_BY(acm_crit_sect_);
bool first_10ms_data_ GUARDED_BY(acm_crit_sect_);
int last_encode_value_ GUARDED_BY(acm_crit_sect_);
CriticalSectionWrapper* callback_crit_sect_;
AudioPacketizationCallback* packetization_callback_