Skip to content

GitLab

E
ESP32-audioI2S
Unverified Commit 77118582 authored by Wolle's avatar Wolle Committed by GitHub
Browse files
Options

stream detection

parent e70dd869
Hide whitespace changes
Inline Side-by-side
Showing with 14 additions and 71 deletions
src/Audio.cpp
View file @ 77118582
......@@ -3,8 +3,8 @@
*
* Created on: Oct 26.2018
*
* Version 2.0.7b
* Updated on: Nov 28.2022
* Version 2.0.7c
* Updated on: Nov 29.2022
* Author: Wolle (schreibfaul1)
*
*/
......@@ -2950,12 +2950,9 @@ void Audio::processWebStreamTS() {
const uint16_t maxFrameSize = InBuff.getMaxBlockSize(); // every mp3/aac frame is not bigger
uint32_t availableBytes; // available bytes in stream
static bool f_tmr_1s;
static bool f_stream; // first audio data received
static bool f_firstPacket;
static uint32_t byteCounter; // count received data
static uint32_t tmr_1s; // timer 1 sec
static uint32_t loopCnt; // count loops if clientbuffer is empty
static uint8_t ts_packet[188]; // m3u8 transport stream is 188 bytes long
uint8_t ts_packetStart = 0;
uint8_t ts_packetLength = 0;
......@@ -2969,8 +2966,6 @@ void Audio::processWebStreamTS() {
f_firstPacket = true;
byteCounter = 0;
chunkSize = 0;
loopCnt = 0;
tmr_1s = millis();
m_t0 = millis();
ts_packetPtr = 0;
m_controlCounter = 0;
......@@ -3029,34 +3024,10 @@ void Audio::processWebStreamTS() {
}
// timer, triggers every second - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
if((tmr_1s + 1000) < millis()) {
f_tmr_1s = true; // flag will be set every second for one loop only
tmr_1s = millis();
}
// if the buffer is often almost empty issue a warning - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
if(InBuff.bufferFilled() < maxFrameSize && f_stream){
static uint8_t cnt_slow = 0;
cnt_slow ++;
if(f_tmr_1s) {
if(cnt_slow > 50 && audio_info) audio_info("slow stream, dropouts are possible");
f_tmr_1s = false;
cnt_slow = 0;
}
}
// if the buffer can't filled for several seconds try a new connection - - - - - - - - - - - - - - - - - - - - - -
if(f_stream && !availableBytes){
loopCnt++;
if(loopCnt > 200000) { // wait several seconds
loopCnt = 0;
AUDIO_INFO("Stream lost -> try new connection");
httpPrint(m_lastHost);
return;
}
// if the buffer is often almost empty issue a warning - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
if(f_stream){
if(streamDetection(availableBytes)) return;
}
if(availableBytes) loopCnt = 0;
// buffer fill routine - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
if(true) { // statement has no effect
......@@ -3073,7 +3044,7 @@ void Audio::processWebStreamTS() {
if(f_stream){
static uint8_t cnt = 0;
cnt++;
if(cnt == 6){playAudioData(); cnt = 0;}
if(cnt == 3){playAudioData(); cnt = 0;} // aac only
}
return;
}
......@@ -3083,13 +3054,10 @@ void Audio::processWebStreamHLS() {
const uint16_t maxFrameSize = InBuff.getMaxBlockSize(); // every mp3/aac frame is not bigger
const uint16_t ID3BuffSize = 1024;
uint32_t availableBytes; // available bytes in stream
static bool f_tmr_1s;
static bool f_stream; // first audio data received
static bool firstBytes;
static uint32_t byteCounter; // count received data
static size_t chunkSize = 0;
static uint32_t tmr_1s; // timer 1 sec
static uint32_t loopCnt; // count loops if clientbuffer is empty
static uint16_t ID3WritePtr;
static uint16_t ID3ReadPtr;
static uint8_t* ID3Buff;
......@@ -3099,10 +3067,8 @@ void Audio::processWebStreamHLS() {
f_stream = false;
byteCounter = 0;
chunkSize = 0;
loopCnt = 0;
ID3WritePtr = 0;
ID3ReadPtr = 0;
tmr_1s = millis();
m_t0 = millis();
m_f_firstCall = false;
firstBytes = true;
......@@ -3164,34 +3130,10 @@ void Audio::processWebStreamHLS() {
}
}
// timer, triggers every second - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
if((tmr_1s + 1000) < millis()) {
f_tmr_1s = true; // flag will be set every second for one loop only
tmr_1s = millis();
}
// if the buffer is often almost empty issue a warning - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
if(InBuff.bufferFilled() < maxFrameSize && f_stream){
static uint8_t cnt_slow = 0;
cnt_slow ++;
if(f_tmr_1s) {
if(cnt_slow > 25 && audio_info) audio_info("slow stream, dropouts are possible");
f_tmr_1s = false;
cnt_slow = 0;
}
}
// if the buffer can't filled for several seconds try a new connection - - - - - - - - - - - - - - - - - - - - - -
if(f_stream && !availableBytes){
loopCnt++;
if(loopCnt > 200000) { // wait several seconds
loopCnt = 0;
AUDIO_INFO("Stream lost -> try new connection");
httpPrint(m_lastHost);
return;
}
// if the buffer is often almost empty issue a warning or try a new connection - - - - - - - - - - - - - - - - - - -
if(f_stream){
if(streamDetection(availableBytes)) return;
}
if(availableBytes) loopCnt = 0;
if(InBuff.bufferFilled() > maxFrameSize && !f_stream) { // waiting for buffer filled
f_stream = true; // ready to play the audio data
......@@ -3204,7 +3146,7 @@ void Audio::processWebStreamHLS() {
if(f_stream){
static uint8_t cnt = 0;
cnt++;
if(cnt == 1){playAudioData(); cnt = 0;}
if(cnt == 3){playAudioData(); cnt = 0;} // aac only
}
return;
}
......@@ -4898,13 +4840,14 @@ boolean Audio::streamDetection(uint32_t bytesAvail){
cnt_slow = 0;
}
if(InBuff.bufferFilled() < InBuff.getMaxBlockSize()) cnt_slow++;
if(InBuff.bufferFilled() > InBuff.getMaxBlockSize() * 2) return false; // enough data available to play
// if no audio data is received within three seconds, a new connection attempt is started.
if(bytesAvail) {tmr_lost = millis() + 1000; cnt_lost = 0;}
if(tmr_lost < millis()){ // 3s no data?
if(tmr_lost < millis()){
cnt_lost++;
tmr_lost = millis() + 1000;
if(cnt_lost == 3){
if(cnt_lost == 5){ // 5s no data?
cnt_lost = 0;
AUDIO_INFO("Stream lost -> try new connection");
connecttohost(m_lastHost);
......
src/Audio.h
View file @ 77118582
......@@ -2,7 +2,7 @@
* Audio.h
*
* Created on: Oct 28,2018
* Updated on: Nov 28,2022
* Updated on: Nov 29,2022
* Author: Wolle (schreibfaul1)
*/
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment